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| 03:11.48 | hypknight|Away | is now away - Reason : Auto-Away (Away from Keyboard for 60 minutes) |
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| 06:16.33 | linocisco | hi all |
| 06:17.07 | linocisco | http://linoxide.com/tools/install-setup-asterisk-13-pbx-centos-7/ shows how to install asterisk13 on CentOS7. is it correct? is MariaDB required? |
| 06:23.57 | tacky | required? no |
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| 06:29.00 | linocisco | tacky, is certified asterisk free to use? |
| 06:38.15 | tacky | it is free software (euphemism: open source) |
| 06:39.07 | tacky | u can get commercial support pretty easily though, if u so desire |
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| 06:42.31 | linocisco | tacky, i see now. thanks for answer |
| 06:42.32 | tacky | i dunno what "certified" asterisk means though. whenever i hear of certification, all i can think of is... https://twitter.com/asscert |
| 06:43.00 | [TK]D-Fender | Certified = meaningless |
| 06:43.04 | tacky | linocisco: no prob, i'm a n00b myself |
| 06:43.09 | [TK]D-Fender | unless you have an actual support contract it changes nothing |
| 06:43.49 | tacky | and even then |
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| 06:44.45 | tacky | certification is for people who need props, on either end of the counter |
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| 08:15.50 | ntt | Hi, when I click on "apply config" in freepbx I have an error: "Unable to connect to remote asterisk. Unable to connect to asterisk through the CLI in file /var/lib/asterisk/bin/retrieve_conf on line.24". If I do "amportal stop; amportal start;" the problem is solved. Someone can help me please? |
| 08:22.01 | tacky | ntt: you lost me at 'click'. i don't use a gui. no clue. sorry. |
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| 08:24.40 | linocisco | hi all |
| 08:25.43 | linocisco | instead of downloading pjsip2.4.5 which is latest. I have downloaded 2.4 and compiled and asterisk is also compiled. how can I install pjsip latest version? |
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| 09:28.58 | linocisco | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 09:29.01 | linocisco | http://pastie.org/private/rffskoev0oojp1ww7tw1w |
| 09:29.09 | linocisco | what do I do? |
| 09:30.18 | linocisco | Nov 23 15:54:58 pbx.linocisco.local systemd[1]: asterisk.service: Supervising process 29990 which is not our child. We'l...exits |
| 09:30.39 | linocisco | what should Ido? |
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| 10:14.23 | juned | <PROTECTED> |
| 10:14.40 | juned | How you're connecting to asterisk ? is it running ? |
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| 11:10.34 | mjamilfarooq | hi folk |
| 11:10.37 | mjamilfarooq | s |
| 11:10.55 | mjamilfarooq | i am confused about asterisk support for video IVR |
| 11:11.05 | mjamilfarooq | anyone can help me ? |
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| 11:19.05 | _omer | How to know if .tiff file that I am using to SendFax(), is valid for asterisk? |
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| 11:37.00 | mjamilfarooq | hey there |
| 11:37.08 | mjamilfarooq | anyone for help |
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| 12:55.31 | Parham | Hi. When I am dialing another Asterisk system from my Asterisk machine, I get a "busy/conjested" error. How can I debug the problem and find the root cause? When I turn on SIP debugging, I see a "401 Unauthorized" error. |
| 12:56.06 | stephanj | is allowguests enabled? |
| 12:56.48 | Parham | No, but I have added an entry to sip.conf. |
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| 13:02.24 | sgelardi | Hello, I have been struggling with this for a few days: I have a SBC interconnecting SIP Trunks to Asterisk. I want to associate incoming calls -> Asterisk to a particular trunk depending on destination user, via header contents, or perhaps some other parameters. The problem seems though that Asterisk can only do it by source IP Address, which in this case will always be the SBC. Is this a limitation or are there other parameter |
| 13:04.06 | Parham | stephanj: No, but the Asterisk servers both have been added to each other's sip.conf. |
| 13:04.20 | stephanj | mh okey |
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| 13:05.06 | Parham | stephanj: On one server, when I type 'sip show peers' I can see the other server, and it has 'OK (1 ms)' in front of it. However, on the other server (which is doing the actual dial command), the other server is noted as 'unmonitored'. |
| 13:05.41 | stephanj | was just a oneshot idea :) |
| 13:05.47 | stephanj | got no real clue about asterisk |
| 13:07.17 | sgelardi | stephenj: in your trunk configuration do you have qualify=yes on one system but not on the other? |
| 13:07.43 | Parham | sgelardi: Let me look... |
| 13:10.12 | sgelardi | stephanj: I missed your original quesiton, but I am assuming by the last few posts, that you are simply just trying to connect two asterisk systems together via a trunk right? |
| 13:10.36 | Parham | sgelardi: Yes. |
| 13:10.40 | stephanj | sgelardi: Parham has the problem :) |
| 13:11.04 | sgelardi | multitasking........ :) |
| 13:11.07 | Parham | sgelardi: And I have qualify=yes on one side but not on the other, yep. |
| 13:11.43 | sgelardi | Parham: that is why it shows unmonitored, basically it means the one where you have it set is sending SIP Options to the other one. |
| 13:12.32 | Parham | sgelardi: Ah. So it has nothing to do with the "401 Unauthorized" error? |
| 13:13.38 | Parham | sgelardi: The story is that I am trying to create a trunk between two virtual machines, and when one of them calls the other, I get a "busy/conjested" error. |
| 13:14.01 | sgelardi | parham: correct it just means you are not monitoring the status of that trunk |
| 13:14.25 | sgelardi | parham: and in the other direction? |
| 13:14.33 | Parham | sgelardi: I haven't tried that. |
| 13:14.52 | Parham | sgelardi: And thanks for the explanation re 'qualified=yes'. |
| 13:15.35 | sgelardi | parham: yw, 401 Unauthorized is a challenge, meaning the one system you probably have to authenticate against |
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| 13:16.30 | sgelardi | parham: are you just doing this to learn asterisk or, what are your goals exactly? |
| 13:18.07 | Parham | sgelardi: We have 2 Asterisk systems in production. One of them is in our local site, which our customers call. However, we have to have another Asterisk at the bank site that does the actual payment. There is a trunk between these two systems, and at some point, the call is forwarded to the bank asterisk. The bank asterisk gets the credit card info, does the payment, then forwards the call back. |
| 13:18.43 | Parham | sgelardi: Now, since these two systems are in production, I am trying to make a change. I thought the best way to do this is to create two development versions locally with Vagrant, connect them together, and play around. |
| 13:19.09 | sgelardi | parham: I agree its not good to touch a production system |
| 13:19.29 | sgelardi | parham: so what do you have in your lab? sip phone --- asterisk ----- asterisk ---- sip phone? |
| 13:20.50 | Parham | sgelardi: I'm new to Asterisk, so excuse me if I amke a mistake, but I think it's landline phone -> access erver -> asterisk -> asterisk. |
| 13:21.39 | Parham | sgelardi: Access server* |
| 13:21.40 | sgelardi | parham: ok, and the access server is what exactly? Does it communicate with asterisk via SIP? |
| 13:21.49 | Parham | sgelardi: Yep. |
| 13:23.27 | sgelardi | parham: and attached to the other Asterisk? |
| 13:23.59 | Parham | sgelardi: Yes. It's more like 32 Asterisk though. |
| 13:24.21 | Parham | sgelardi: So, we have 32 local asterisks that connect to 4 bank asterisks through a trunk. |
| 13:25.53 | sgelardi | parham: ok, but you are just working with 2 right now, so can you maybe just layout your call flow? You have to figure out where your 401 is coming from |
| 13:26.42 | sgelardi | parham: maybe it comes from asterisk, maybe it comes from the gateway, or another asterisk system that is interconnecting. In a chat it is obviously very difficult to guide you without very specific infomraiton, so try tracing your problem to the source. |
| 13:27.17 | Parham | sgelardi: Ah! |
| 13:27.33 | Parham | sgelardi: To be specific, I have soft phone -> asterisk -> asterisk. |
| 13:27.52 | Parham | sgelardi: All of these are on my computer. The two Asterisk machines are virtual machines. |
| 13:28.09 | Parham | sgelardi: The 401 unauthorized error is happening when the first machine tries to dial the second machine. |
| 13:28.32 | sgelardi | parham: ok but what is on the other side of the second Asterisk? do you have a single sip phone w/ 2 lines? |
| 13:29.44 | Parham | sgelardi: Ah no, there are just extensions that play some audio files, ask some information, run some AGI scripts, and then trunk the call back. |
| 13:30.52 | Parham | sgelardi: The two Asterisk systems don't actually connect the calls to any operators. These are systems that are used for providing services through phone calls. Does that make sense? I feel like I'm overcomplicating things? :) |
| 13:38.29 | sgelardi | parham: so you expect to make a call like this: sip phone ---- asterisk ------ asterisk --- auto-attendent |
| 13:38.56 | Parham | sgelardi: Exactly! |
| 13:39.15 | Parham | sgelardi: I have to leave the workplace. Is it possible to continue the conversation tomorrow? |
| 13:47.28 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 13:47.45 | sgelardi | ello, I have been struggling with this for a few days: I have a SBC interconnecting SIP Trunks to Asterisk. I want to associate incoming calls -> Asterisk to a particular trunk depending on destination user, via header contents, or perhaps some other parameters. The problem seems though that Asterisk can only do it by source IP Address, which in this case will always be the SBC. Is this a limitation or are there other parameters |
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| 14:08.32 | [TK]D-Fender | sgelardi: Asterisk is NOT a proxy. The call is coming from an * server with whatever auth it is sending. that server itself is not a proxy and not forwarding auth from the phone that caused it to dial either |
| 14:09.51 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:12.06 | [TK]D-Fender | You simply need to fix your auth issues between the servers |
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| 14:29.21 | linocisco | hi all |
| 14:29.35 | linocisco | <PROTECTED> |
| 14:33.36 | sgelardi | D-Fender: were you trying to answer my question or my help with my correspondence w/ parham?? If so he left along time ago |
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| 14:35.41 | WIMPy | linocisco: And what's the answer? |
| 14:36.18 | *** join/#asterisk DavidHuebner (~hubie@167.206.156.240) |
| 14:36.26 | linocisco | WIMPy, i am looking for answer using asterisk -vcccc. not found myself. |
| 14:36.34 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 14:36.58 | WIMPy | You posted a question, but that was to you. What;s the answer? |
| 14:37.03 | [TK]D-Fender | You shouldn't be starting * manually like that |
| 14:38.17 | [TK]D-Fender | sgelardi: I somewhat mixed your 2 issues |
| 14:38.24 | linocisco | [TK]D-Fender, I am starting * with asterisk -rvvv. but i got that error |
| 14:38.27 | WIMPy | And that command surely won't give the message you posted. |
| 14:38.43 | WIMPy | Did you read that error? |
| 14:38.55 | linocisco | WIMPy, yes. |
| 14:38.55 | WIMPy | Did you notice that it contains a question? |
| 14:40.12 | linocisco | WIMPy, yes. I tried to locate that file and verified asterisk.conf for run directory |
| 14:40.42 | WIMPy | And did you find it? |
| 14:40.45 | linocisco | WIMPy, i found it is in correct path |
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| 14:40.53 | WIMPy | Great. |
| 14:40.56 | WIMPy | What about permissions? |
| 14:41.25 | WIMPy | And have you checked if asterisk is actually running? |
| 14:43.15 | linocisco | WIMPy, pls see http://pastie.org/private/6twgwuys94f08i0z1ixtjq |
| 14:45.34 | WIMPy | Don't know what that is, but it doesn't confirm Asterisk is running. Just that it tries to start it a few times which could be an indication of some issue. |
| 14:45.58 | sgelardi | D-Fender: Alright, no advise for my incoming trunk question huh? |
| 14:46.46 | linocisco | WIMPy, so what should I do? |
| 14:46.49 | [TK]D-Fender | sgelardi: is your SBC doing it's own auth to *? |
| 14:47.20 | WIMPy | ps aux|grep asterisk |
| 14:47.48 | [TK]D-Fender | linocisco: Where do we see that file as existing and with a value it in it? |
| 14:49.11 | sgelardi | D-Fender: no the SBC, in short the SBC from Asterisks POV is a stateful proxy. Asterisk will register through the SBC but the SBC is not doing any authenitication |
| 14:49.30 | [TK]D-Fender | sgelardi: So it is a pure proxy? |
| 14:49.39 | sgelardi | no its a BBUA |
| 14:49.55 | sgelardi | but for instance |
| 14:50.03 | sgelardi | VIA headers from WAN -> LAN are kept intact |
| 14:50.23 | sgelardi | but in the other direction they are not (hiding the LAN from the WAN) |
| 14:50.38 | [TK]D-Fender | sgelardi: So it isn't forwarding the packets, it is a re-write. Well it'll only auth as a single entity then. With chan_sip you havea dilaplan function to inspect the headers from the invite so you can make a decision based on them |
| 14:50.40 | sgelardi | but unlike a proxy the contact header will always have the SBC's IP |
| 14:50.44 | [TK]D-Fender | but the call will look like it's just the SBC |
| 14:50.53 | [TK]D-Fender | "core show functions like SIP" |
| 14:51.00 | linocisco | [TK]D-Fender, ok.http://pastie.org/private/jqfydo0sotzbzahmx4teg |
| 14:52.26 | linocisco | brb |
| 14:54.21 | sgelardi | D-Fender: so basically there are no parameters that can accomplish what I am looking for, Asterisk by default will only look at layer 3 information to associate it with a trunk? |
| 14:59.08 | igcewieling | sgelardi: you can't auth by IP or by username / secret? |
| 14:59.20 | [TK]D-Fender | * auths based on username with "type=friend", or IP "type=peer" |
| 14:59.34 | [TK]D-Fender | how it challenges them depends on "insecure=" being set or not |
| 14:59.40 | igcewieling | [TK]D-Fender: not complicated 8-) |
| 15:00.23 | [TK]D-Fender | sgelardi: You could create distinct entires of type=friend with the username and no auth required. |
| 15:00.40 | [TK]D-Fender | sgelardi: And just restrict the IP's to your SBC |
| 15:01.04 | [TK]D-Fender | heads into a meeting for a while |
| 15:01.36 | sgelardi | igcewieling: I thought username was only used for outbound calls? |
| 15:02.13 | sgelardi | igcewiling: plus I am pretty sure that the username does not show in the Invite of inbound clals |
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| 15:04.00 | sgelardi | D-Fender: for all outgoing calls I need a username / password and already have that defined, so on incoming calls it will classify the call with the username as a criteria? Taken I use the type friend? |
| 15:05.02 | sgelardi | D-Fender: Which header is it looking at with username=xxxxxxx? I am assuming the Request-URI? |
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| 16:13.28 | igcewieling | sgelardi: try the from uri. |
| 16:22.44 | sgelardi | D-Fender/igcewieling: i do not think type=friend will work, nor the from URI, I think the call direction is not clear. The call direction is PSTN ---- SBC ----- Asterisk |
| 16:23.14 | sgelardi | therefore using the from header will change on a per call basis |
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| 16:33.19 | igcewieling | sgelardi: exactly. you are back to auth by IP |
| 16:39.26 | sgelardi | igcwieling: unfortunitelly doesnt work either because i have multiple SIP trunks that come through the SBC that need to be handled differently |
| 16:41.32 | sgelardi | so really I guess the point is clear, asterisk is limited in this respect that there is no built in functionality, therefore i should create a seperate incoming trunk from the SBC with a custom context that further identifies which trunk it really belongs to |
| 16:43.23 | igcewieling | When you have such limitations there is no good solution. |
| 16:44.14 | igcewieling | I've never had that sort of problem. Our SBC talks to multiple carriers and the calls seem to work our Asterisk boxes just fine. |
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| 17:56.05 | sgelardi | igcewieling: do you do any number manipulation on the SBC itself? How would you handle doing all normalization on Asterisk on a per trunk basis? |
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| 18:09.04 | igcewieling | sgelardi: Not sure what you mean. Call comes into Asterisk from our SBC, Asterisk know where calls to that TN are supposed to go and then Asterisk sends the call to the endpoint. |
| 18:09.46 | igcewieling | I only care about the carrier when we send calls out. |
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| 18:10.46 | igcewieling | When the call comes in, I normalize DNIS and ANI to 1NXXNXXXXXX before going any further in the dialplan. |
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| 18:37.05 | dorphalsig | Hey |
| 18:38.03 | dorphalsig | I'm having trouble getting a WSS phone to work. I can register, but when I try to call the device I get a SIP/2.0 488 Not Acceptable Here |
| 18:38.15 | dorphalsig | as a response |
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| 19:25.50 | sgelardi | dorphalsig: check your codecs |
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| 19:35.56 | dorphalsig | Hey, anybody around today? |
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| 20:12.29 | dorphalsig | Hey.. My internet seems to be coming and going. I'll ask again (sorry). I'm trying to call an RTC web phone (SIPJS and SIPML) and Im getting a 488 response |
| 20:13.05 | dorphalsig | All I'm seeing in the logs is a notice: 2015-11-21 20:11:53] NOTICE[68343][C-00000006]: chan_sip.c:10043 process_sdp: Received AVP profile in audio answer but AVPF is enabled, disabling: audio 8000 RTP/AVP 3 110 8 0 97 101 |
| 20:13.28 | igcewieling | dorphalsig: that looks like a message you might be able to google |
| 20:15.10 | dorphalsig | igcewieling: But is that enough to get me a rejected call? |
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| 20:20.40 | igcewieling | dorphalsig: I have no idea, but my first google hit looked promising. |
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| 20:22.24 | dorphalsig | igcewieling: the webphone extension has the AVP/AVPF/Encryption = yes. The other extension (regular sofpthone) has all of those turned off, which is how it should be, right? |
| 20:26.59 | *** part/#asterisk saint_ (~saint_@104.156.228.147) |
| 20:32.04 | dorphalsig | igcewieling: So, the AVP notice is gone. Now all I have left is the 488 :( |
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| 20:40.31 | igcewieling | sorry, I'm not crazy enough to try WebRTC |
| 20:55.01 | justdave | is there a way to specify an exclusion for localnet? i.e. I have a /8 network which is local -- except for one /22 network inside it. |
| 20:55.40 | justdave | is there a way to say "not this /22" or do I have to do a whole bunch of localnet lines listing all of the largest possible subnets to cover the rest of that network? |
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| 21:12.32 | igcewieling | first shoot your network designer. |
| 21:13.13 | igcewieling | justdave: have you checked the sip.conf.sample file? There might be something useful in there. As far as I know Asterisk cannot do what you want, but I'm not sure of that. |
| 21:20.18 | justdave | it's the guest wifi network |
| 21:20.29 | justdave | doesn't have access to anything else in the network, but can get to the internet. |
| 21:20.58 | justdave | but people on that wifi are having trouble using softphone clients and they work fine on the real internet or from the actualy corporate wifi network |
| 21:21.37 | justdave | I was guessing because the clients think they have IP addresses that are inside that main /8 that asterisk was trusting them even though they're coming from the external NAT IP by the time it sees them |
| 21:22.44 | justdave | for now I'm just telling people "you have a softphone login that means you also have a login for the real wifi, just use it" :) |
| 21:23.35 | WIMPy | Don't you think that most clients behind NAT would come from IPs you listed as localnet? |
| 21:25.27 | justdave | I would suspect that in a few cases. Most people seem to be in 192.168/16 instead of 10/8 though |
| 21:25.35 | justdave | probably something weird with the firewall |
| 21:25.49 | justdave | was just thinking that might be an easy thing to test |
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| 23:33.05 | dorphalsig | hey |
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