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| 01:08.45 | smkelly | file: YES |
| 01:08.58 | smkelly | ducks |
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| 05:20.51 | Raden | how can i match inbound area code to route calls ? |
| 05:21.04 | Raden | so lets say 630 area code goes to sip/1000 ? |
| 05:24.12 | WIMPy | I'm not sure if you can use patterns in the callerID part of extensions. |
| 05:25.10 | WIMPy | But you can always use *If or an AGI or just goto the part of the callerID or ise it to fetch data from a DB or .... |
| 05:28.50 | Raden | there no way to match like an outbound call context ? |
| 05:29.45 | WIMPy | All calls are incomming. |
| 05:31.11 | Raden | yes |
| 06:23.16 | MaliutaLap | WIMPy: use AEL if (CALLERID(NUM) = <regex>) .... |
| 06:25.23 | WIMPy | MaliutaLap: So you are the one using AEL? ;-) |
| 06:34.21 | MaliutaLap | WIMPy: pretty sure I'm not the only one :P |
| 06:34.51 | MaliutaLap | WIMPy: could be worse - I could be using LUA ;) |
| 06:35.16 | WIMPy | Well, the thing is that using LUA does at least make sense. |
| 06:36.09 | MaliutaLap | not really |
| 06:36.14 | MaliutaLap | AEL is fine |
| 06:36.35 | WIMPy | But AEL is only translated to standard dialplan. |
| 06:37.07 | MaliutaLap | Still lets me programmaticly express my dialplan |
| 06:37.27 | WIMPy | But it doesn't give you additional features. |
| 06:37.39 | WIMPy | Probably even less. |
| 06:38.15 | MaliutaLap | I haven't come across a feature that it didn't support |
| 06:38.31 | MaliutaLap | are you saying LUA adds extra features? |
| 06:39.54 | WIMPy | Well, it adds LUA. |
| 06:40.12 | WIMPy | It's not just a translation. |
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| 08:08.06 | gavimobile | it seems all but one of my sip trunks are having audio issues, is there a way I can verify if the issue is actually with this one provider and not a local pbx issue? |
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| 08:19.58 | kafal | hello, I am trying to setup confbridge in Asterik 12.3. Agent is logged in and I am able to make and outbound cal. But I am not able to put both the calls in the same conference bridge. I keep getting following message in the logs "No channel type registered for 'Agent'". Is this error related to the issue I am facing? Any ideas how I can fix this issue? |
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| 09:17.45 | WIMPy | What do agents have to do with conferences? |
| 09:35.11 | mic_ | are there any known issues when SIP provider is using cirpack apart from the "keepalive" that google shouts about? |
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| 09:51.45 | kafal | Hi WIMPy: is there any sample examples to show how to put to users in a conference? |
| 09:52.04 | kafal | for asterisk 12.3 |
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| 12:02.54 | asterisco | someone use click to call with hp+asterisk? |
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| 13:53.50 | gusto | so |
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| 16:13.17 | joesmoe | i have two trunks to the same provider with seperate usernames but their calls keep getting routed to the same place |
| 16:13.19 | joesmoe | i.e. incoming calls from one accounts are being routed to both incoming rules |
| 16:19.03 | [TK]D-Fender | Show us the call |
| 16:19.12 | [TK]D-Fender | ? cli |
| 16:19.18 | [TK]D-Fender | "sip set debug on" |
| 16:19.21 | [TK]D-Fender | oops... |
| 16:19.31 | [TK]D-Fender | (at * CLI) |
| 16:19.48 | [TK]D-Fender | and "both incoming rules" does not make sense as worded. |
| 16:20.24 | joesmoe | So i have an incoming call rule that mathes localphone.com Trunk1 and another that matches all calls for localphone.com Trunk2 |
| 16:20.36 | joesmoe | but when i call a DID @ localphontTrunk1, it matches loaclphone.com trunk2 |
| 16:20.38 | joesmoe | doing a debug now |
| 16:21.17 | joesmoe | i'm using digium |
| 16:21.20 | joesmoe | not the cli... :( |
| 16:21.21 | [TK]D-Fender | include both sip.conf entries masking only the secret |
| 16:22.07 | [TK]D-Fender | [12:21]joesmoei'm using digium [12:21]joesmoenot the cli... <- what is this supposed to mean? |
| 16:22.19 | joesmoe | i think it may be because i have the insecure mode on the trunks set to 'very' |
| 16:22.25 | joesmoe | digium is like a gui for asterisk |
| 16:22.41 | [TK]D-Fender | The GUI has been dead for over 3 years. |
| 16:22.54 | [TK]D-Fender | more like 4 at this point |
| 16:23.15 | [TK]D-Fender | GUI also doesn't factor into this. |
| 16:23.17 | joesmoe | when using insecure=very, and having 2 trunks going to the same provider, coudl that be whats messing things up? |
| 16:23.27 | [TK]D-Fender | It's a show the peers |
| 16:23.32 | [TK]D-Fender | show the peers |
| 16:25.31 | joesmoe | even when removing the trunk i'm still getting my PBX when making a call to one of the incoming numbers |
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| 16:52.41 | jermy | I'm having issues getting BLFs to update in a timely manner on a Yealink phone. They update fine when it re-subscribes the SIP connection, so I assume some notifications aren't working? (Asterisk version 11.10.2) |
| 16:54.22 | WIMPy | I do also have issues with subscriptions soptting to work after some time. It works again for a time if I either restart the phone or Asterisk. |
| 16:55.03 | WIMPy | I should try to get the time, but I have a feeling that it works for longer if I restart both at the same time. |
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| 17:08.15 | Apteryx | Hi! It seems I fail to understand what direcmedia=nonat is supposed to do. I have a peer, named FPL (sip trunk, registered) sitting outside my nat at address sip:208.65.240.44. My Asterisk is portforwarded to port 5060 inside my nat at address 192.168.0.10. I'm connecting to it from a local peer, maxim-VostroV130, using 192.168.0.10. When FPL tries to call maxim-VostroV130, Asterisk causes a re-invite. This doesn't work because maxim-VostroV13 |
| 17:10.21 | [TK]D-Fender | as both should be listed as "nat=no" they should both think they're allowed to reinvite which is bad. |
| 17:10.34 | [TK]D-Fender | You're far better off just preventing them across the board. |
| 17:10.49 | [TK]D-Fender | Otherwise you're going to be playing cat & mouse with a lot of pain |
| 17:11.52 | *** join/#asterisk seanbrig1t (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 17:12.30 | Apteryx | [TK]D-Fender: Hello! Given that I set localnet to 192.168.0.0 and that this is a NATed network (because this address is inherently private), shouldn't Asterisk be smart and detects a peer with address 192.168.0.100 as NATed too? |
| 17:12.46 | Apteryx | localnet=192.168.0.0/24 to be exact. |
| 17:13.08 | [TK]D-Fender | Apteryx: there is no detection. These settings are explicit |
| 17:13.17 | [TK]D-Fender | and that is what you put in your peer |
| 17:13.20 | [TK]D-Fender | "nat=" |
| 17:13.55 | SeRi | Penguin: got the mikrotik going. |
| 17:14.15 | Apteryx | [TK]D-Fender: Ok! I guess that explains the behaviour I'm seeing! Thanks. |
| 17:14.43 | SeRi | Penguin: complicated when you dont know what is going on. once I learn a few triks from other folks it was easy from there |
| 17:16.03 | [TK]D-Fender | Apteryx: You're welcome |
| 17:16.56 | Apteryx | :) |
| 17:38.06 | Apteryx | Strange. After putting "directmedia=nonat" and "nat=force_rport,comedia" in my NATed peers definition, Asterisk is still re-invitting my NATed peers with external one. Confused. |
| 17:38.24 | *** join/#asterisk derekb (~derekb@d24-57-14-108.home.cgocable.net) |
| 17:38.51 | WIMPy | Glad to hear that I'm not the only one where it doesn;t seem to work. |
| 17:39.12 | derekb | hi there, been having an issue with contexts on sip trunks for a little bit now. basically had an issue when my pstn gateway passed a call back to asterisk that it was making a loop due to using the wrong context |
| 17:39.27 | derekb | ive changed to the correct context on the sip trunk, but now it seems outbound dialing is effected. |
| 17:39.51 | derekb | heres the output from the asterisk CLI when i try to make an outbound call from the customers phone system to asterisk: http://pastebin.com/SBiV6Jty |
| 17:40.05 | [TK]D-Fender | Outbound from your other devices has nothing to do with a context on the peer yo are dialing |
| 17:40.22 | derekb | Haha, yes we had this discussion yesterday |
| 17:40.35 | [TK]D-Fender | And this is STILL the same broken Goto |
| 17:40.41 | [TK]D-Fender | this is not a SIP problem |
| 17:41.20 | [TK]D-Fender | that also does not look like you dialing out.. that looks like you dialing IN from your gateway |
| 17:41.41 | derekb | its an IP phone, registered to a customers telephone system. placing a call out to 5195517056 |
| 17:41.54 | Apteryx | WIMPy: Ok. Thanks for sharing. Maybe a bug in the version of Asterisk we are using (mine: 11.7.0 from Ubuntu 14.04 repo) |
| 17:42.06 | [TK]D-Fender | well look where that phone's peer is pointing. |
| 17:42.07 | derekb | via sip trunk SIP/5199977432 using an outbound CLID of 5197376668 |
| 17:42.15 | [TK]D-Fender | Why is the phone pointing to from-outside? |
| 17:42.19 | derekb | i agree it looks like an inbound call for some reason |
| 17:42.30 | WIMPy | Apteryx: If it's a bug, it's not new. |
| 17:42.31 | [TK]D-Fender | that is where you peer to the gateway shold be pointed, not peers for your phones |
| 17:43.14 | derekb | what do you mean? |
| 17:44.23 | [TK]D-Fender | <PROTECTED> |
| 17:44.36 | derekb | yes |
| 17:44.43 | [TK]D-Fender | SIP/5199977432 <- if this is a PHONE then you should not have pointed it to that context |
| 17:44.52 | derekb | thats a trunk |
| 17:44.55 | derekb | toa phone system, from asterisk |
| 17:44.56 | [TK]D-Fender | that peer is PHONE peer, not your GATEWAY peer, correct? |
| 17:45.04 | [TK]D-Fender | ... |
| 17:45.09 | [TK]D-Fender | There is no "trunk to a phone" |
| 17:45.13 | [TK]D-Fender | you are mising ins * outs |
| 17:45.18 | [TK]D-Fender | mixing* |
| 17:45.32 | derekb | i didnt say it wasa a trunk to a phone :( |
| 17:45.35 | [TK]D-Fender | .... |
| 17:45.37 | WIMPy | heard there are no trunks at all with SIP |
| 17:45.52 | [TK]D-Fender | derekb: Get clear, fast. |
| 17:46.01 | derekb | sorry |
| 17:46.03 | [TK]D-Fender | WHAT is on the other side of that peer? |
| 17:46.21 | derekb | softphone was makign the call. |
| 17:46.33 | [TK]D-Fender | You just said gateway. |
| 17:46.39 | [TK]D-Fender | Now it's a softphopne? |
| 17:46.43 | derekb | *sigh* |
| 17:46.48 | derekb | i am confused, im sorry |
| 17:46.52 | derekb | i appreciate you trying to help |
| 17:49.55 | [TK]D-Fender | what device is directly coming in to * there? [5199977432] is the peer being matched in sip.conf. What is the exact device making that comm? |
| 17:51.28 | derekb | Sorry [TK]D-Fender, I feel like an idiot here. THis is the exact call setup. Softphone logged into customer PBX, makign call out. Have trunk setup between phone system and asterisk, which is the SIP/5199977432 peer. |
| 17:51.57 | [TK]D-Fender | "customer PBX" < what is this |
| 17:51.59 | *** join/#asterisk camelCase (~camelCase@unaffiliated/camelcase) |
| 17:52.00 | [TK]D-Fender | Youa re being vague |
| 17:52.26 | [TK]D-Fender | Some other type of system? |
| 17:52.43 | derekb | yeah its an NEC telephone system |
| 17:52.47 | derekb | not asterisk |
| 17:52.52 | MaliutaLap | shudders |
| 17:52.55 | [TK]D-Fender | From your description Asterisk precisely NOTHING about the softphone. It is only talking to that other PBX |
| 17:53.09 | [TK]D-Fender | The fact there is something beyond it isn't *'s concern |
| 17:53.17 | [TK]D-Fender | So * ONLY talks to that other PBX? |
| 17:54.00 | derekb | what are you referring to as *? |
| 17:54.05 | [TK]D-Fender | * = asterisk |
| 17:54.21 | MaliutaLap | [TK]D-Fender: tautology! |
| 17:54.23 | derekb | duh |
| 17:54.24 | [TK]D-Fender | You know.. the key on your keyboard. That's what it's called. |
| 17:54.32 | MaliutaLap | :) |
| 17:54.40 | derekb | sorry :( |
| 17:54.58 | [TK]D-Fender | http://www.asterisk.org/ <- you might recognize it from their giant logo. |
| 17:55.06 | camelCase | anyone here mess with a cosmos(ess7) to SIP gateway using *? |
| 17:55.09 | derekb | asterisk is talking to other pbx's as well, not just that one. |
| 17:55.35 | MaliutaLap | camelCase: * is my sip gateway :) |
| 17:55.42 | camelCase | no shit |
| 17:55.54 | camelCase | have you ever use * to interface with ESS7? |
| 17:56.13 | [TK]D-Fender | derekb: Ok, so if [5199977432] is supposed to be an account for the PBX to call and dial OUT... then this is probably supposed tro be treated as a PHONE in your * system, and that is the wrong context for that peer. |
| 17:56.15 | [TK]D-Fender | Go fix it |
| 17:56.25 | derekb | gotcha. |
| 17:56.37 | MaliutaLap | camelCase: nope. all my phones talk to * which talks to an ITSP for external purposes |
| 17:56.47 | derekb | [TK]D-Fender: thanks for helping, and being patient |
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| 18:16.25 | derekb | [TK]D-Fender: looks like things are working :D |
| 18:16.46 | derekb | [TK]D-Fender: thank you so much for the tips, and being patient. i really appreciate the assistance. |
| 18:17.05 | [TK]D-Fender | You're welcome. |
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| 18:25.24 | [TK]D-Fender | ~best |
| 18:25.27 | infobot | best for what? please define what you mean by "best" Gloria Gaynor! Tina Turner! Aretha Franklin! Men without Hats! Women without Hats! Flock of Seagulls!, or fvwm! Women without clothes! |
| 18:25.35 | [TK]D-Fender | raspberrypifan: ^ |
| 18:26.07 | raspberrypifan | when did come out with women without hats |
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| 18:46.38 | WIMPy | camelCase: What is _E_SS7? |
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| 19:03.44 | camelCase | http://en.wikipedia.org/wiki/Signalling_System_No._7 |
| 19:30.02 | WIMPy | camelCase: I don't find any E there, just regular SS7. |
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| 19:50.40 | camelCase | ESS is another way it is called, like ESS5 |
| 19:50.56 | TazzNZ | morning all |
| 19:52.43 | WIMPy | Not seen that before. And google doesn't seem to know much about it, either. |
| 19:53.04 | camelCase | it's what we called it in the early 1990's when it was coming out |
| 19:53.19 | camelCase | you may find old text files named ESS7 |
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| 20:03.50 | *** join/#asterisk igcewieling1 (~ewieling@ip98-183-26-100.pn.at.cox.net) |
| 20:04.27 | igcewieling1 | Does anyone have experience running Asterisk as non-root and binding to a low port, such as 443, for TLS? I tried it using setcap (aka libcap) but I must be doing something wrong. |
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| 21:40.08 | Apteryx | I am right that the safest (and simple to deploy) way to deal with a PSTN gateway is to find a SIP `trunk` provider supporting TLS connections? I know that the PSTN part of the call will not be encrypted, but this is still better security than no security... and ZRTP is not possible in this scenario. |
| 21:40.56 | WIMPy | Are you talking about a gateway or an ITSP? |
| 21:41.13 | Apteryx | Gateway |
| 21:41.37 | WIMPy | What's the point of encrypting on your own network? |
| 21:42.13 | Apteryx | Well, sorry, I would encryp both. |
| 21:42.15 | WIMPy | And the NSA will save a copy of the call anyway. |
| 21:42.25 | WIMPy | Both what? |
| 21:42.53 | Apteryx | By SIP `trunk` provider I meant ITSP (not familiar with the right terminology yet) |
| 21:43.33 | Apteryx | So I would provide TLS on my * personal gateway |
| 21:43.39 | [TK]D-Fender | " PSTN Gateway" implies a device you own hooked to standard telco lines. |
| 21:43.45 | Apteryx | And I would setup the ITSP peer to use TLS as well. |
| 21:43.53 | [TK]D-Fender | "ITSP" is a service provider over the internet |
| 21:43.58 | [TK]D-Fender | do not mix these up |
| 21:44.33 | [TK]D-Fender | are we talking about hardware, or an ITSP? |
| 21:44.49 | Apteryx | ITSP. Sorry for mixing up. |
| 21:47.13 | Apteryx | So to rephrase it right, I plan to offer TLS connections to my * PXB, and connect * to an ITSP over TLS. That way my calls would be encrypted all the way until the PSTN network (assuming the ITSP doesn't screw it up) |
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| 21:49.19 | [TK]D-Fender | that's one way |
| 21:50.59 | Apteryx | [TK]D-Fender: Would another way be better in your opinion? |
| 21:51.21 | [TK]D-Fender | You could direct VPN to yor provider if they offer |
| 21:53.56 | MaliutaLap | VPN has stronger encryption too |
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| 21:55.01 | Apteryx | MaliutaLap: OK. Thanks. |
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| 22:25.53 | Apteryx | MaliutaLap: I'm not sure we can fully trust VPN though... I remember reading this: http://marc.info/?l=openbsd-tech&m=129236621626462&w=2 a few years back, and then this: https://highdeserttechs.com/technology-resources/security-and-privacy-news/can-we-trust-ipsec.html. Of course it is a bit pointless to talk about NSA backdoors when the finality will be hitting the PSTN, but still, I understand TLS is simpler thus safer. |
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| 22:49.21 | igcewieling1 | Does anyone have experience running Asterisk as non-root and binding to a low port, such as 443, for TLS? I tried it using setcap (aka libcap) but I must be doing something wrong. |
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| 23:24.36 | ipengineer | is the present of rport=5060 an indication of SIP ALG or some kind of manipulation usually? |
| 23:25.12 | ipengineer | Trying to trace down a problem that happens on a particular network and that is about the only thing I see different on the INVITE |
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