| 01:04.45 | tm1000 | file: wiki down :-( https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification |
| 01:07.57 | file | Rusty may be doing maintenance |
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| 01:49.13 | nickgaw | Hi, What would the cost be for someone to help me write the configuration files for a two extention asterisk system with voicemail and where all calls are recorded with each caller on different channel of a stereo file and for a short message prompting the user to dial the extention they wish to call? |
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| 06:13.38 | Primer | Hi, is there a PPA for ubuntu 16/mint 18? I can't seem to find one. Asterisk segs on start on a freshly installed mint 18 machine |
| 06:22.38 | *** join/#asterisk CeBe (~CeBe@a81-14-235-90.net-htp.de) |
| 06:28.35 | wyoung | Primer: a PPA for what? |
| 06:28.42 | wyoung | asterisk? |
| 06:29.00 | Primer | yes |
| 06:29.04 | wyoung | ummm no there doesn't need to be one as it exists in the repos already |
| 06:29.27 | wyoung | I don't know about mint 18 but Ubuntu 14.04 works fine, I havent tested 16.04 though |
| 06:29.39 | Primer | I did a more conservative install and now it's no longer crashing |
| 06:29.45 | wyoung | ah |
| 06:30.55 | Primer | But it's still not working, but I haven't looked at any config changes required between what was in 14.04 and 16.04 |
| 06:31.04 | Primer | ubuntu versions...I thing it was 11 |
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| 06:31.22 | Primer | sorry, asterisk version 11 on 14.04 to 13 on 16.04 |
| 06:32.00 | wyoung | ah ok |
| 06:33.36 | Primer | [Aug 8 06:33:10] ERROR[11508]: message.c:1388 ast_msg_tech_unregister: No 'sip' message technology found. |
| 06:33.41 | Primer | scratches head... |
| 06:33.56 | wyoung | Primer: filesystem permissions on /etc/asterisk/sip.conf |
| 06:34.10 | Primer | yup, that was it |
| 06:35.13 | wyoung | has happened to me a few times :) |
| 06:35.16 | Primer | and it's all working |
| 06:35.17 | Primer | thanks |
| 06:35.20 | wyoung | any time |
| 06:35.49 | wyoung | Primer: it would be great if that is the first error message :) |
| 06:35.57 | wyoung | might submit a bug report |
| 06:36.08 | Primer | that would make things much easier to fix |
| 06:36.22 | wyoung | I agree |
| 06:36.37 | Primer | Now to see if all runs well from systemd... |
| 06:36.53 | Primer | I'll refrain from giving my opinion regarding systemd |
| 06:37.55 | Primer | ok, everything looks good...I can die happy now |
| 06:38.09 | Primer | wyoung: thanks again |
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| 06:38.56 | wyoung | ah systemd |
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| 06:48.49 | *** join/#asterisk Haris (~haris@unaffiliated/haris) |
| 06:50.10 | Haris | ok. what tutorial do I follow to install asterisk for ws/wss video calls + cdr + vm + etc etc ? |
| 06:50.37 | Haris | I have centos 7 64bit fresh installed for this. What version of asterisk do I install on it ? |
| 06:51.19 | Haris | I was watching a few yrs old youtube video which said asterisk 11, 12 didn't have good support for ws/wss calls (at the time of making that video) |
| 06:51.45 | Haris | Correction: ..time of making of+ that video) |
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| 07:00.13 | wyoung | Haris: yum install asterisk is probably your best bet unless you wnt to compile from source |
| 07:00.32 | wyoung | centos has a habbit of being slow to adopt the latest versions of software |
| 07:01.00 | Haris | is there a custom yum repo its on ? its not in base, updates, epel |
| 07:01.13 | wyoung | ummmm, it's not? |
| 07:02.06 | Haris | re-checking |
| 07:02.30 | wyoung | asterisk.x86_64 : The Open Source PBX |
| 07:02.46 | wyoung | yeah it's there |
| 07:03.21 | wyoung | I have the standard repo + epel |
| 07:03.28 | wyoung | not sure which one it is in though |
| 07:03.47 | wyoung | I am using Centos 6.6 though |
| 07:04.08 | Haris | I'm on 7.x latest |
| 07:04.16 | wyoung | it should be in there then |
| 07:04.32 | wyoung | if not you can compile it from source, there is probably srpm for it somewhere too |
| 07:04.35 | Haris | yum info asterisk should show which repo its on |
| 07:05.22 | wyoung | It's in base, epel, extras and updates |
| 07:05.48 | wyoung | 1.8.32.3 though, it is a somewhat dated release |
| 07:06.16 | Haris | its not in repo for 7.x |
| 07:06.31 | Haris | checking another mirror |
| 07:07.40 | Haris | its not in base for 7 |
| 07:08.41 | Haris | not in base, extras, updates for 7.x. checking epel |
| 07:09.16 | wyoung | dang |
| 07:09.40 | wyoung | I just do a sudo apt-get install asterisk :) |
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| 07:10.49 | Haris | not in epel as well |
| 07:12.30 | Haris | ok. so I was using freepbx as a way to configure asterisk before now. found out its not a good option for my work. is there another tool which is great at providing front end for configuring asterisk, rather than needing me to poke around files |
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| 07:24.02 | wyoung | Haris: Not off the top of my head. Most people just edit files. |
| 07:24.15 | Haris | hmm |
| 07:24.18 | wyoung | you can create a web frontend if you want to |
| 07:31.18 | Haris | hmm |
| 07:39.52 | wyoung | do it |
| 07:40.08 | Haris | http://packages.asterisk.org/centos/centos-asterisk-13.repo |
| 07:40.16 | wyoung | woooo! |
| 07:41.00 | Haris | checking if its been updated for c7 |
| 07:41.59 | Haris | not updated for 7 |
| 07:42.01 | Haris | :| |
| 07:42.45 | wyoung | :( |
| 07:43.01 | wyoung | Install Ubuntu / Debian? :) |
| 07:43.35 | *** join/#asterisk FarhaadN (~Farhad@82.99.206.194) |
| 07:45.20 | Haris | ok. with debian I can also try freeswitch. probably a good idea |
| 07:46.07 | FarhaadN | i have a problem, randomly extention shows inuse in core show hints |
| 07:46.26 | FarhaadN | and any extention monitor this extention ,show show inuse |
| 07:46.37 | Haris | is ws/wss calls support been turned into separate mods on debian pkgs' ? |
| 07:46.45 | wyoung | Haris: or compile from source |
| 07:46.48 | wyoung | that is what I usually do |
| 07:49.19 | Haris | can srpms for 6 be built on 7 ? |
| 07:49.32 | wyoung | depends on the dependencies |
| 07:49.50 | wyoung | source code <3 |
| 07:50.21 | Haris | http://packages.asterisk.org/centos/6/asterisk-13/SRPMS/asterisk-13.3.2-1_centos6.src.rpm is dated 23-Oct-2015 14:10 Size: 7.8M |
| 07:50.35 | Haris | pretty old ? |
| 07:51.38 | wyoung | build from source or use another distro or use asterisk's yum repo or all of the above |
| 07:52.21 | Haris | asterisk hasn't been updated to support c7 yet. will have to come down to c6 |
| 07:52.24 | FarhaadN | wyoung: can u help me? |
| 07:54.42 | wyoung | FarhaadN: ummm I am only computer qualified though so depends on your issues ;) |
| 07:54.47 | HRH_H_Crab | hi all, i am using asterisk on debian testing with the distro versions. i think i *may* be seeing this bug: https://issues.asterisk.org/jira/browse/ASTERISK-22564 |
| 07:55.00 | wyoung | FarhaadN: I have never experienced that issue before |
| 07:55.05 | HRH_H_Crab | symptoms are that when i make outgoing calls, the recipient can hear me, but i cannot hear them. |
| 07:55.17 | HRH_H_Crab | incoming calls work fine (with directmedia=no) |
| 07:55.22 | wyoung | FarhaadN: however check your firewall / logs, you may be under attack |
| 07:55.26 | HRH_H_Crab | note that i connect to my upstream sip provider using ipv6 |
| 07:55.39 | HRH_H_Crab | but my handsets connect using ipv4 to my dual stack asterisk server. |
| 07:55.40 | wyoung | ok that sounds like a firewall issue |
| 07:55.48 | wyoung | is your asterisk server behind NAT? |
| 07:55.52 | HRH_H_Crab | no. |
| 07:55.58 | HRH_H_Crab | there is no nat involved. |
| 07:56.05 | wyoung | you have a firewall? |
| 07:56.07 | HRH_H_Crab | it connects to the upstream provider using ipv6 (so no nat) |
| 07:56.10 | HRH_H_Crab | yes |
| 07:56.20 | HRH_H_Crab | but there is no nat between the handsets and asterisk |
| 07:56.24 | wyoung | ipv6 means no NAT? |
| 07:56.37 | wyoung | ipv6 means no NAT?/ |
| 07:56.39 | wyoung | what about a firewall though |
| 07:56.41 | HRH_H_Crab | wyoung: not necessarily, but in this case there is no nat. |
| 07:56.48 | HRH_H_Crab | i also note that if i enable the hep server |
| 07:56.54 | HRH_H_Crab | when making *outbound* calls |
| 07:56.57 | HRH_H_Crab | (not inbound) |
| 07:57.07 | HRH_H_Crab | i see errors about "address family mismatch" |
| 07:57.14 | FarhaadN | wyoung: there are'nt any firewall ,this is local |
| 07:57.17 | wyoung | if you can't hear stuff then it is usually a RTP issue |
| 07:57.23 | wyoung | in particular, being blocked |
| 07:57.26 | HRH_H_Crab | wyoung: i think so |
| 07:57.35 | HRH_H_Crab | but in this case i think that its not the firewall blocking anything |
| 07:57.54 | HRH_H_Crab | i think its the dual stack server not correctly bridging ip6 / ipv4 as per that bug report i referenced. |
| 07:58.11 | HRH_H_Crab | i note he has an excerpt of sip debug included |
| 07:58.32 | HRH_H_Crab | but im not sure reading that, how he has diagnosed the problem |
| 07:58.47 | wyoung | ok well I have never used IPv6 |
| 07:59.19 | HRH_H_Crab | apparently (from his report) he only sees the problem in asterisk 12. im using 13.8 |
| 07:59.24 | HRH_H_Crab | he says that version 11 works fine |
| 08:00.02 | HRH_H_Crab | and i have heard from someone else using a dual stack asterisk server bridging ipv4 / ipv6 works, so i may try that to confirm |
| 08:00.19 | HRH_H_Crab | but i thought i would mention it here unless anyone knew about this problem. |
| 08:06.49 | wyoung | <PROTECTED> |
| 08:06.50 | wyoung | :( |
| 08:06.52 | wyoung | TK would know |
| 08:07.08 | wyoung | [TK]D-Fender: *bump* |
| 08:11.27 | Haris | centos or debian for asterisk's best support ? |
| 08:13.43 | Haris | with debian I had a slight issue with pkg re-installs. it would cache already installed pkgs, rather than re-installing them, which in my case would generate pkg version conflicts. I prefer centos for clarity of pkg installs .. especially in case of re-installs. This happens for me when I manually re-install asterisk |
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| 08:15.03 | Haris | manual install tutorial for asterisk doesn't mention the fact that node js server needs to be setup. is node js server needed for ws/wss calls ? |
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| 08:44.58 | davlefou | hi, is it possible to send and receive sms via asterisk when we used voip line? |
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| 08:57.37 | wyoung | Haris: for best support buy a digium PBX |
| 09:22.19 | Haris | well, I'm not implementing a traditional voice pbx per se |
| 09:22.38 | Haris | this one is for webrtc calls over ws/wss over the Internet mostly. no pstn connectivity at all |
| 09:24.03 | wyoung | and? |
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| 10:21.51 | Haris | ? |
| 10:25.19 | Haris | looks like community has stopped maintaining or upgrading pkgs for new OS version(s) or for centos 7 |
| 10:25.30 | Haris | or there is a break in it |
| 10:34.25 | Samot | Haris: Actually explain your problem. |
| 10:34.59 | WIMPy | He did. He istalled from packages. |
| 10:35.05 | Samot | Ugh. |
| 10:37.18 | Samot | WIMPy: Before you go down this rabbit hole, [TK]D-Fender and I have spent almost the last 8 weeks hand holding Haris on this endeavor. |
| 10:37.54 | WIMPy | I actually missed that. |
| 10:38.04 | Samot | WIMPy: Total lack of the ability to follow instructions or absorb the knowledge of things discussed, at length, with him. |
| 10:38.11 | Samot | AND |
| 10:38.30 | Samot | Dove in to VoIP/SIP, Asterisk, everything about the same time we started helping him. |
| 10:41.19 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 10:41.45 | Samot | And you wouldn't have seen most of it. He started with FreePBX but it's design won't let him implement WebRTC they way he would like to. |
| 10:42.19 | Samot | Just a little FYI. |
| 10:42.37 | Samot | file has seen it, he knows the drain of this help vamp. |
| 10:44.19 | file | hmmm? oh |
| 10:47.39 | Samot | Haris> ok. what tutorial do I follow to install asterisk for ws/wss video calls + cdr + vm + etc etc ? <--- You've been provided the links for this a few times. Including last week when I provided them to you. |
| 10:48.35 | WIMPy | Wasn't first, won't be last. |
| 10:49.15 | Samot | Crtl+D generally stops that problem. |
| 10:49.28 | Samot | In most browsers. |
| 10:49.52 | WIMPy | ? |
| 10:50.02 | Samot | Bookmark command. |
| 10:50.13 | Haris | Samot: well, someone also mentioned later on that .. that url was for something else, rather than this |
| 10:50.21 | Samot | Which one? |
| 10:50.26 | Samot | The one for WebRTC? |
| 10:50.28 | WIMPy | Oh, right. |
| 10:50.32 | Samot | That has the links for Secure Calling? |
| 10:50.43 | Haris | yes |
| 10:50.48 | Samot | And links in the Wiki for basic user / device setup. |
| 10:51.13 | Samot | No, the Wiki link to make WebRTC work is for making WebRTC work. |
| 10:51.32 | WIMPy | is currently upgrading openssl so I can upgrade Asterisk tonight. |
| 10:52.16 | Haris | I'm getting my c7 VM re-installed for c6. and the other VM re-installed with debian latest to start from scratch. Looks like asterisk is not yet ready for c7 |
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| 10:52.49 | WIMPy | If distro matters, something is seriousely wrong. |
| 10:53.06 | Samot | It would mean he has to install from source. |
| 10:53.07 | Haris | there's an asterisk yum repo with c6 rpm pkgs. old though. It doesn't have pkgs for c7 |
| 10:54.20 | WIMPy | Yes, but how likely is it that a packaged version does what you want? I find it a little questionable to provide packaged versions of an application as configurable as Asterisk. |
| 10:54.29 | Samot | Haris: I'm going back to my original advice to you for almost the last two months: Hire someone. |
| 10:54.45 | *** join/#asterisk KValchev (~KValchev@ns.atsoftconsult-bg.com) |
| 10:54.49 | Haris | I'v relayed that option to decision makers. let's see what decision is made |
| 10:54.52 | WIMPy | Haris: Have you check if they are new enough to enable you to do what you want? |
| 10:55.41 | Samot | Would a cost analysis of this endeavor thus far help them make their decision. |
| 10:55.47 | Haris | WIMPy: I need browser to browser video calls over ws/wss, between mob app user and bank agent and then cdr + possibly vm |
| 10:55.49 | Samot | Because they are bleeding money on this. |
| 10:56.06 | Samot | Oh christ, now a mobile app is involved. |
| 10:56.31 | Haris | shakes head. why did I have to mention that. now he's going to get more confused :| |
| 10:56.35 | WIMPy | Haris: I didn't as what you want to do. I asked if you checked, if the available version will let you do that. |
| 10:56.50 | Haris | it doesn't have mods for ws as separate pkg |
| 10:57.05 | Samot | WIMPy: Watch the edge of the rabbit hole. |
| 10:57.26 | Haris | I'm going to check it out. and remove it completely if it doesn't work. and then go with install from source .. all over again |
| 10:59.10 | Haris | mob app part is just going to have one button that will initiate sip call with our sip server. that's a no brainer or more like a sip client configured to call one sip server/proxy/exchange. that's not the part which is proving hard to work on. the server part, in which we want to deploy a sip server that'll actually make the calls to happen |
| 10:59.35 | Samot | Yeah, the most vital piece of the service. |
| 10:59.37 | Haris | is the part which requires major attention |
| 10:59.43 | Samot | You know, making the calls work. |
| 10:59.59 | Haris | =) |
| 11:00.06 | Samot | It's not funny Haris. |
| 11:00.11 | Samot | Really, it's not. |
| 11:00.18 | WIMPy | .me has given up on that a long time ago. |
| 11:00.19 | Haris | I'm not laughing. that was a smile in agreement |
| 11:00.39 | Samot | You're being paid to do a project that you are totally dependant on IRC community support to do. |
| 11:00.48 | Haris | I'm not a contractor |
| 11:00.52 | Haris | or on a part time job |
| 11:00.54 | Samot | Are you getting paid? |
| 11:01.02 | Haris | my position is full time. |
| 11:01.06 | Samot | So? |
| 11:01.12 | Samot | That means you are being paid to do this. |
| 11:01.19 | Samot | "Make WebRTC calls work" |
| 11:01.23 | Samot | That's your job right now. |
| 11:01.33 | Samot | They are paying you to get Asterisk and WebRTC calls working. |
| 11:01.35 | Samot | You are in here. |
| 11:02.04 | Samot | Asking every step of the way "How do I do X?" |
| 11:03.11 | Haris | yes. that's because there isn't a clear way to doing X or my practical experience has shown that the documentation does not lead to a working solution or doesnt' lead to a working solution in my specific condition |
| 11:03.28 | Samot | Your practical experience is asking everything. |
| 11:03.35 | Haris | in apache, I can load/unload mods to have or not have functionality |
| 11:03.44 | Samot | You can do the same in Asterisk. |
| 11:03.56 | Samot | You can load and unload modules you need/don't need. |
| 11:04.18 | Haris | so far, with asterisk when I enable web sockets mod, I can't register properly with asterisk with a browser on a random laptop on the same LAN segment |
| 11:04.22 | Samot | Continue to compare apples to oranges though. |
| 11:04.45 | WIMPy | Probably because of NAT. |
| 11:04.56 | Haris | nat does not apply on the same lan segment |
| 11:05.04 | Samot | Something he's been told numerous times as well. |
| 11:05.23 | WIMPy | That doesn't mean it's causing trouble with SIP. |
| 11:05.34 | Samot | You've installed the SSL certs right? |
| 11:05.42 | Haris | yes |
| 11:05.48 | Haris | I had installed them |
| 11:05.59 | Samot | All the TLS settings are set and correct? |
| 11:06.08 | WIMPy | Some attempts to make SIP work over NAT will make it fail if a NAT situation is deteced but does not apply. |
| 11:06.11 | Samot | All the device settings for transport and TLS are set? |
| 11:06.14 | Haris | I'm re-doing to OS to show my config again |
| 11:06.20 | Haris | on two separate OSs |
| 11:06.52 | Haris | WIMPy: why would anyone need nat for communication between boxes .. on the same LAN segment ? |
| 11:07.05 | Haris | OSs = VMs |
| 11:07.58 | file | if it's not working then reinstalling or trying a different OS isn't going to help - you have ot investigate and undestand *why* it's not working |
| 11:08.18 | WIMPy | Haris: That's not what I said. The fact that it's there, even if somewhere else, it can do harm with SIP. |
| 11:08.58 | file | in the case of websockets that's narrowing things down - looking at wireshark captures to see the negotiation |
| 11:09.13 | WIMPy | That's just the classic case of a bugfix intrducing new bugs. |
| 11:09.52 | file | websockets themselves haven't been touched and are used quite heavily - as ARI also uses them for the event system |
| 11:10.56 | file | and WebRTC itself can work, but when it doesn't it's hellish to debug because there's lots of moving parts to it |
| 11:12.03 | Haris | and since its happening in browser. there's no errors or logs made |
| 11:12.31 | file | there's the Javascript console, and you can also enable logging within Chrome, as well there's chrome://webrtc-internals |
| 11:12.40 | Haris | hmm |
| 11:12.49 | file | if anyone is seriously wanting to get into WebRTC and make it a product that is stuff they HAVE to learn along with WebRTC itself |
| 11:12.58 | file | otherwise you pay someone to deal with all of it |
| 11:12.58 | Haris | when I input my sip uri in sip client, I get nothing. just the page reloads "as was" |
| 11:13.37 | Haris | if chrome://webrtc-internals helps debug or give me errors, I'll at least know which direction to look into |
| 11:13.51 | Haris | or where the problem lies in the path |
| 11:16.42 | Samot | file: No one has correlated that even once they get it working, they have to be able to support it. |
| 11:16.47 | Haris | I need to be able to run a manual sip session via telnet. if I can succeed in that, I can try to build a tool, a basic tool, say in VB, which exchanges basic sip headers to troubleshoot basic connectivity between client <-> server |
| 11:16.56 | Samot | Stop. |
| 11:16.59 | Samot | Just stop. |
| 11:17.12 | Samot | You don't even understand the basics of SIP and WebRTC.. |
| 11:17.20 | Samot | Don't talk about making tools for SIP testing. |
| 11:18.41 | Haris | I'v read on sip and webrtc. seen tutorials on webrtc on youtube. seen sip client web apps. I know SIP, RTP, RTCP, SRTP |
| 11:18.48 | Samot | You do? |
| 11:18.58 | Samot | By watching a few youtube videos? |
| 11:19.16 | Samot | Half comprehension of some Wiki pages? |
| 11:19.39 | Haris | asterisk is not the only topic where I read or learned about sip, etc etc |
| 11:19.50 | Haris | I'll leave it at that |
| 11:19.56 | Samot | Haris: If you had a basic understanding of any of that.... |
| 11:20.03 | Samot | We wouldn't be two months into this. |
| 11:20.20 | Samot | We wouldn't have had to explain every step of TLS calling. |
| 11:20.22 | Haris | basic understanding of a protocol cannot be compared to implementing a product |
| 11:20.30 | Samot | You wouldn't be asking HALF the questions you ask. |
| 11:20.33 | Haris | a protocol and a product are two different things |
| 11:20.41 | Samot | ...... |
| 11:21.01 | WIMPy | Samot: You need to get more productive. |
| 11:21.09 | Samot | I know. |
| 11:21.12 | Haris | we have run webrtc calls directly between two browsers |
| 11:21.21 | Samot | I've already installed two servers while doing this. |
| 11:21.23 | Haris | just not able to get it work through a client/server architecture yet |
| 11:21.29 | Samot | Configured FOP2 on one of them already. |
| 11:21.44 | Samot | Haris: You get that is the BASICS?! |
| 11:21.49 | WIMPy | Oh, does that still exist? |
| 11:21.57 | Samot | So what you are saying is "I can't get the basics to work" |
| 11:21.59 | Samot | But once I do... |
| 11:22.02 | Samot | Stand back! |
| 11:22.18 | Samot | WIMPy: Yup. |
| 11:22.29 | Samot | Deploy it a lot. |
| 11:22.52 | WIMPy | Is it still SIP only? |
| 11:23.24 | Samot | Yeah, I don't think it has IAX support. |
| 11:23.27 | Haris | I did get asterisk to work. I did get microsip <--asterisk--> microsip video calls to work both over udp and tls |
| 11:23.39 | Haris | just unable to configure asterisk for ws transport based calls |
| 11:23.47 | Samot | That's the basics of a standard SIP call. |
| 11:23.59 | Samot | We're talking about the basics of a WebRTC call. |
| 11:24.32 | WIMPy | It would need at least DAHDI support. Although I'm not using that any more, either. |
| 11:24.35 | Haris | we'v already done a basic laptop 1 (browser) <---> laptop 2 (browser) video call over ws. it works with no asterisk in bwtween |
| 11:24.38 | Haris | between+ |
| 11:24.57 | Samot | The bottom line is this: As long as the steps in a Wiki or video that you follow result in success, you're fine. If they result in failure, you're completely lost as to why. |
| 11:25.45 | Haris | that's because the client does not throw errors and the server either doesn't get any traffic or isn't configured with verbose setting enough to give debug output for the activity I'm performing |
| 11:25.57 | Haris | hate browsers for not throwing errors :| |
| 11:26.41 | Samot | If it's something like JS that is client side based... |
| 11:26.47 | WIMPy | too. But it's probably a good thing, as no professional web pagge has less than 100 errors. |
| 11:26.50 | Haris | with apache, smtp, imap, I can perform a basic connectivity test via telnet, exchange headers. see if the server side responds. with sip, I don't know how to do that yet |
| 11:26.54 | Samot | The Developer Tools with it's console and error reporting will show it. |
| 11:27.03 | Samot | If it's sever side, then the server will show it. |
| 11:27.06 | Haris | will try that |
| 11:27.48 | Samot | Haris: apples and oranges. |
| 11:28.53 | Haris | products are ofcourse apples and oranges. but the same method of troubleshooting can apply to sip, asterisk |
| 11:30.09 | Samot | No, comparing SMTP to SIP. |
| 11:30.32 | Haris | s/products/protocols/g |
| 11:30.58 | Haris | cool feature |
| 11:34.51 | Haris | there are http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing |
| 11:34.56 | Haris | I just don't know how to work them yet |
| 11:35.34 | Haris | these tools are for basic voice calls |
| 11:37.11 | Haris | I'm a CCVP |
| 11:40.59 | *** join/#asterisk FarhaadN (~Farhad@82.99.206.194) |
| 11:41.16 | *** join/#asterisk Zogot (~Adium@185.21.52.255) |
| 11:41.47 | Haris | restarting irc client to change language :| |
| 11:41.48 | *** part/#asterisk Haris (~haris@unaffiliated/haris) |
| 11:42.01 | Samot | Just no idea. |
| 11:43.13 | *** join/#asterisk Haris (~haris@unaffiliated/haris) |
| 11:56.53 | *** join/#asterisk nohitall (~nohitall@unaffiliated/nohitall) |
| 11:57.57 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 11:58.00 | *** join/#asterisk chuckf (~chuckf@pool-108-45-91-234.washdc.fios.verizon.net) |
| 11:58.13 | nohitall | hi, i got a weird isue with incoming calls, 4/5 times the call fails, logs say invite fails, not authorized, so I googled that and read about allowing guest sip calls but that doesnt make sense to me since in 1/5 it works, it works randomly. I am a asterisk noob though, anybody can point me in the right direction? |
| 12:00.18 | WIMPy | Calls from an ITSP? |
| 12:04.07 | nohitall | some mobile |
| 12:04.20 | nohitall | I am not familiar with the terms of SIp stuff sorry |
| 12:04.30 | nohitall | test calls from a mobile phone into office, randomly works |
| 12:04.35 | WIMPy | ~itsp |
| 12:04.35 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 12:05.20 | nohitall | in logs I see "Using INVITE request as basis...SIP/2.0 401 Unauthorized" |
| 12:05.21 | WIMPy | How are those calls comming in to Asterik? |
| 12:06.29 | nohitall | uh via blueface the provider? |
| 12:07.54 | WIMPy | Then I guess they have servers on 5 different IPs and you only configured one of them. |
| 12:08.03 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 12:08.11 | *** join/#asterisk puzzola (~puzzola@unaffiliated/puzzola) |
| 12:09.00 | nohitall | i should see in the log which IP it comes from right? |
| 12:09.32 | WIMPy | Probably not. |
| 12:09.50 | *** join/#asterisk Tiffon (~name@unaffiliated/tiff0n) |
| 12:10.04 | WIMPy | Or, the security log should have it. |
| 12:10.16 | nohitall | under which settings panel do I see this for the IPs? |
| 12:10.36 | nohitall | ah sorry, config file |
| 12:10.40 | WIMPy | What software are you talking about? |
| 12:10.43 | nohitall | im using elastix |
| 12:10.50 | nohitall | but I can also directly check the confs, works too |
| 12:10.59 | WIMPy | Ouch. |
| 12:11.04 | WIMPy | sip.conf |
| 12:11.27 | nohitall | this is first day I ever came across this, so im doing crashcourse learning here, so forgive my missing knowledge :D |
| 12:11.37 | WIMPy | If your ITSP uses multiple IPs, you need a peer per IP of your ITSP. |
| 12:11.46 | nohitall | well sip.conf is empty |
| 12:12.07 | nohitall | does freepbx store the settings somewhere else? |
| 12:12.08 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-kxenhcowdbtgxesy) |
| 12:12.37 | WIMPy | No idea. And no support for 3rd party GUIs in here. |
| 12:13.15 | nohitall | whats the setting called, so I can google int he docs |
| 12:13.20 | WIMPy | And not sure how much support you get in #freepbx when you mention elastix. |
| 12:13.30 | WIMPy | There's not setting. |
| 12:13.47 | WIMPy | It's about setting up peers. |
| 12:13.56 | nohitall | you said it would be in sip.conf, how would it look like |
| 12:14.08 | TandyUK | its not a setting, its an entire config block in sip.conf, per peer |
| 12:14.21 | nohitall | yea I get that |
| 12:14.23 | WIMPy | Look for a sip.conf.sample |
| 12:14.53 | nohitall | k thanks |
| 12:17.08 | nohitall | whats the overall opinion of elastix here? :D |
| 12:23.52 | nohitall | the peer details for trunk are set by domain |
| 12:23.57 | nohitall | so I figure that elimantes IP issue? |
| 12:24.04 | WIMPy | ~esatix |
| 12:24.09 | nohitall | host=sip.domain.tld |
| 12:24.21 | WIMPy | Nope. It causes it. |
| 12:24.56 | nohitall | I get 2 IPs on resolve though, not 5 or w/e |
| 12:25.05 | nohitall | so you think I gotta set them both and that should be it? |
| 12:25.16 | WIMPy | That's been bad luck then, I guess. |
| 12:25.23 | WIMPy | yes |
| 12:25.31 | nohitall | yea well 50/50 still can end up with fail/fail/fail/fail/fail/win |
| 12:25.34 | nohitall | ok thanks |
| 12:25.56 | nohitall | another noob question: they'd be both in the same trunk though right? |
| 12:26.14 | nohitall | I assume I can't put 2 host=ip...? |
| 12:26.37 | WIMPy | "trunk" doesn't really mean anything, except as a marketing term. |
| 12:26.42 | WIMPy | Correct. |
| 12:27.25 | nohitall | so I change host=sip.blueface.ie to 2 lines with host=ip1 and host=ip2 |
| 12:27.54 | WIMPy | Correct: You CAN't. |
| 12:27.59 | WIMPy | You have to make two peers. |
| 12:28.10 | nohitall | so 2 trunks |
| 12:28.45 | nohitall | ok how sure can we be this is the reason anyway? if it resolves by IP and I get authorization issue, that means there must be somewhere a esttings that only allows for certain IP correct? |
| 12:29.20 | nohitall | or why is using the domain the issue? |
| 12:30.20 | WIMPy | What do you think how a call is matched to a peer? |
| 12:30.56 | WIMPy | ITSP won't provide credentials when they send you a call. Or at least I haven't seen any do so. |
| 12:31.16 | WIMPy | Which is pretty bad, off course, but well... |
| 12:31.24 | nohitall | yea but if asterisk says "unathorized" against what list does it authorize? |
| 12:31.30 | nohitall | or what reference |
| 12:31.46 | nohitall | if only 1 of the 2 Ips work where is this set? |
| 12:31.52 | WIMPy | The list of configured peers. |
| 12:32.22 | nohitall | in which conf file do I find that? |
| 12:32.30 | WIMPy | If you use host=domain it will chose a random IP for that domain when it parses the config. |
| 12:32.37 | WIMPy | sip.conf |
| 12:33.11 | WIMPy | Or maybe in a database. But I won't dive into "realtime" configuration. |
| 12:33.43 | nohitall | hm but its not there, hence its not set it seems |
| 12:34.11 | nohitall | but the peer details have the domain |
| 12:34.31 | nohitall | host=domain is in peer details |
| 12:34.44 | nohitall | where is the other IP that is not authorized defined then? |
| 12:35.18 | WIMPy | If it was defined it probably wouldn't be unauthorized. |
| 12:35.49 | nohitall | <PROTECTED> |
| 12:35.56 | nohitall | but if incoming callis from the second IP is fails |
| 12:36.06 | *** join/#asterisk Oatmeal (~Suzeanne@99-103-96-198.lightspeed.iplsin.sbcglobal.net) |
| 12:36.23 | WIMPy | Th' IS the way chan_sip works. |
| 12:36.31 | nohitall | ah ok now I get it then :) |
| 12:36.47 | WIMPy | That* |
| 12:36.55 | nohitall | can I do this? host=ip1&ip2&ip3 |
| 12:36.58 | nohitall | just saw that somewhere |
| 12:37.11 | WIMPy | No |
| 12:37.37 | WIMPy | I already told you you need to define one peer per IP. |
| 12:38.19 | nohitall | can tehy have the same settings? othewrise, here stuff like qualify,insecure is set |
| 12:38.27 | nohitall | can port be the same? |
| 12:38.30 | WIMPy | With chan_pjsip it can be handled more efficiently, I think. |
| 12:38.37 | WIMPy | Yes. |
| 12:38.42 | WIMPy | Yes. |
| 12:38.52 | nohitall | thanks |
| 12:39.04 | nohitall | I give it a try |
| 12:39.08 | nohitall | help is much appreciated |
| 12:39.37 | WIMPy | And if you want to tidy things up there's the template thing. But that concept most probably doesn't exist in any GUI. |
| 12:40.43 | *** join/#asterisk wonderworld (~ww@p50858558.dip0.t-ipconnect.de) |
| 12:41.54 | nohitall | is there some cmdline tool to check how many trunks are up/configured |
| 12:42.01 | nohitall | I set the second one, applied changes |
| 12:42.04 | nohitall | just wonder if its set |
| 12:42.28 | WIMPy | 'sip show peers' |
| 12:44.44 | nohitall | do I need to setup second route too?because it says that |
| 12:44.53 | nohitall | trunk is unused, route needs to be set |
| 12:45.28 | WIMPy | Is that a outgoing thing? |
| 12:45.36 | nohitall | yea seems so |
| 12:46.23 | nohitall | ah nvm I can set trunk sequence |
| 12:48.17 | nohitall | well seems still issue still though |
| 12:48.53 | nohitall | 7 fails, all with SIP/2.0 401 Unauthorized |
| 12:49.06 | WIMPy | Do the calls come from the expected IPs? |
| 12:50.24 | nohitall | yea |
| 12:51.35 | nohitall | why is it doing Using INVITE request as basis request |
| 12:51.49 | WIMPy | What? |
| 12:52.10 | nohitall | well the failures all show this |
| 12:52.30 | nohitall | wait I copy to pastebin |
| 12:52.40 | WIMPy | That's how SIP works. |
| 12:53.00 | nohitall | http://pastebin.com/G0MkU7D2 |
| 12:53.59 | WIMPy | "no matching peer" |
| 12:54.38 | nohitall | well but sometimes it works |
| 12:55.04 | WIMPy | Then you need to find the difference. |
| 12:55.42 | nohitall | what does the no matching peer means |
| 12:55.52 | WIMPy | Either at the *CLI with a lot of verbose and debug or with sngrep or wireshark. |
| 12:55.59 | WIMPy | What it says. |
| 12:56.58 | nohitall | i dont even know what to look for |
| 12:57.28 | WIMPy | Sure. |
| 12:57.33 | nohitall | seems setting both trunks by IP caused total failure |
| 12:57.52 | nohitall | or failurerate is much higher now hm |
| 12:58.04 | nohitall | one sec I try to find a succesfull log |
| 12:58.19 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-xogtodhvscvqxnui) |
| 12:58.19 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 12:59.51 | nohitall | WIMPy: works now |
| 12:59.54 | nohitall | WIMPy: THANKS |
| 13:00.04 | nohitall | dont know why seems config reloadwas not instantly |
| 13:00.35 | nohitall | WIMPy: feelfree to query me if you ever need help with something( no matter the topic) |
| 13:02.44 | WIMPy | Errm. That's a pretty broad range. |
| 13:03.07 | nohitall | SIp stuff is the only thing I never worked with |
| 13:03.13 | nohitall | other than that basically anything |
| 13:03.20 | WIMPy | Other Telco stuff? |
| 13:03.23 | nohitall | just saying if u need a favor returned ;) |
| 13:03.47 | WIMPy | has lots of open building sites. |
| 13:04.25 | nohitall | well if you ever get stuck with something maybe I can be of help |
| 13:04.59 | WIMPy | Like do you know anything about Parley service access controllers? |
| 13:06.04 | WIMPy | Or are you in to GNs Lite3000? |
| 13:06.21 | nohitall | I got no experience whatsoever with anything that has to do with telephony :D |
| 13:06.48 | nohitall | so no to both, dont know them |
| 13:07.05 | nohitall | you can ask me for cluster/networking stuff |
| 13:07.08 | WIMPy | So what ARE you good at then? |
| 13:07.23 | WIMPy | Oh. Actually I can. |
| 13:08.01 | WIMPy | For the VoIP stuff I set up TC which used to work great. |
| 13:08.11 | WIMPy | But at some point it stopped doing so. |
| 13:08.35 | WIMPy | And so far I have neither been able to find out why, nor did I find anyone with an idea. |
| 13:10.41 | nohitall | doubt I can help ther either |
| 13:11.24 | nohitall | what you mean with TC? tele closet? |
| 13:11.37 | WIMPy | Trtaffic Control |
| 13:12.37 | nohitall | in what form |
| 13:13.13 | WIMPy | man tc |
| 13:13.23 | WIMPy | Ok, so any other category? |
| 13:15.31 | nohitall | well you can always query and ask, maybe |
| 13:16.01 | WIMPy | Didn't work out too well so far. |
| 13:16.43 | *** part/#asterisk nohitall (~nohitall@unaffiliated/nohitall) |
| 13:16.52 | Haris | ok. preparing my centos vm for asterisk 13 |
| 13:17.00 | Haris | fresh os install |
| 13:33.44 | *** join/#asterisk Spengler (~spengler1@static-96-244-110-2.bltmmd.fios.verizon.net) |
| 13:34.18 | Spengler | is anyone using Asterisk over a VPN? If so are there any caveats? I have been testing it over a 50/50 FIOS connection and it seems to work pretty good. |
| 13:34.41 | Spengler | using a site-to-site VPN, Edgerouter lite from Ubiquiti |
| 13:37.37 | WIMPy | With SIP the usual IP confusions apply. |
| 13:38.15 | Spengler | I had to take the "localnet" configuration out but after I did that everything worked great |
| 13:38.24 | Spengler | audio was crystal clear and no choppiness |
| 13:41.28 | *** join/#asterisk Krikke001 (c13a29fa@gateway/web/freenode/ip.193.58.41.250) |
| 13:42.14 | Krikke001 | Good afternoon, I was wondering if somebody would be able to give me some troubelshooting tips regarding G722/H264 troubleshooting on video calls. |
| 13:52.39 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 13:56.14 | *** join/#asterisk gingitsune (~gingitsun@159.148.5.233) |
| 13:56.30 | gingitsune | What SIP client would recommend for development purposes? |
| 13:57.29 | Samot | To vague of a question. |
| 13:57.49 | gingitsune | Well thats really as specific I can get at this point |
| 13:57.58 | gingitsune | Just shopping around for clients at this point |
| 13:58.08 | gingitsune | Downloaded linphone |
| 13:58.44 | gingitsune | Most of these webpages on softphones seem rather web 1.0. |
| 13:59.32 | WIMPy | You mean they are actually usefull? |
| 14:00.47 | gingitsune | Too early to call on that :) |
| 14:00.55 | gingitsune | I'm sure they are |
| 14:05.25 | *** join/#asterisk eofster (~eofster@pf.eofster.com) |
| 14:09.51 | *** join/#asterisk kharwell (kharwell@nat/digium/x-vvpnlgxoqcvwqunb) |
| 14:21.09 | *** join/#asterisk mike3 (~mike3@162.216.46.127) |
| 14:33.44 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:33.44 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:39.33 | *** join/#asterisk Spengler (~spengler1@static-96-244-110-2.bltmmd.fios.verizon.net) |
| 14:45.11 | *** join/#asterisk chadxz (chadmcelli@nat/digium/x-ajuhfjdynwwkdbax) |
| 14:53.29 | *** join/#asterisk Oatmeal (~Suzeanne@c-68-51-46-116.hsd1.in.comcast.net) |
| 14:55.45 | ntz | [TK]D-Fender: servus |
| 14:56.26 | ntz | i've partially resolved issue I've been asking here about - thanks chan, you've told me crap that delayed me 2 days, anyhow, I have it already solved at my own |
| 14:57.02 | ntz | where do I please find the description and names of available array fields for CALLERID ? |
| 14:57.32 | WIMPy | 'core show function CALLERID' |
| 14:57.49 | WIMPy | Descriptions don't exist. |
| 14:58.55 | ntz | ok, thanks |
| 14:59.17 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-fypbvgaeikqqhzhf) |
| 15:00.15 | WIMPy | If you should find any, tell me. I might then be able to finally tell, which channel(s) do things right and which one(s) don't. |
| 15:01.57 | ntz | WIMPy: http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List |
| 15:02.01 | ntz | this seems usable |
| 15:06.07 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:08.22 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:09.22 | ntz | hmm, I'm still unable to set correctly CID and DID however I have workaround |
| 15:11.14 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-nsmwhtvtmjehtspo) |
| 15:12.04 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:17.15 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:33.00 | ntz | okay, can anybody please help me with following: I have difficulties to set DID and CID between two t38modems .... how do I please dump the environment during the call ? I need to verify, that sending modem at least tries to send something within his CID and DID |
| 15:33.29 | WIMPy | ntz: That may be usefull and, as usual there, outdated. But does ot tell you anything about CALLERID. |
| 15:34.10 | WIMPy | Sending a DID doesn't make sense. |
| 15:34.17 | WIMPy | And for the rest there's the Verbose application. |
| 15:34.46 | WIMPy | What are you trying to do and how does it fail? |
| 15:37.18 | ntz | WIMPy: building a paste for you, gimme a minute |
| 15:41.33 | *** join/#asterisk dym (~patrick@unaffiliated/dym) |
| 15:43.27 | ntz | WIMPy: http://susepaste.org/view/raw/59576492 |
| 15:43.41 | ntz | ^^ please check comments ^### |
| 15:44.26 | ntz | there are only 3 things and full output from `sip set debug on' during sending/receiving fax from 10.0.255.211 to 10.0.255.212 |
| 15:46.05 | ntz | WIMPy: I'd like to achieve nothing lesser than that I see on receiving side both CID and DID |
| 15:47.01 | WIMPy | And what route does the call take? |
| 15:47.16 | WIMPy | AAnd is it about the sending or the receiving side? |
| 15:47.45 | ntz | both |
| 15:47.52 | ntz | this is my actual dialplan: |
| 15:48.10 | WIMPy | Stop posting random snippets. |
| 15:48.26 | WIMPy | I need to get an idea of what your doing first. |
| 15:48.30 | ntz | ok |
| 15:48.43 | WIMPy | What's the full path of the fax call? |
| 15:49.00 | ntz | o.O, I don't understand to the question |
| 15:49.36 | WIMPy | How do you send the fax and what components/technologies are involved on the way to the receiver. |
| 15:49.37 | ntz | there is just t38modem that uses asterisk's sip as an carrier so *everything that comes out is sent at asterisk's sip port |
| 15:49.52 | ntz | t38modem + hylafax |
| 15:50.10 | WIMPy | On both ends? |
| 15:50.17 | ntz | yes, and same versions |
| 15:50.19 | WIMPy | Connected to the same Asterisk? |
| 15:50.23 | ntz | yes |
| 15:50.29 | WIMPy | Good |
| 15:51.31 | ntz | 10.0.255.211 (aka 111211) sends via asterisk (10.0.255.129) to 10.0.255.212 (aka 111212) |
| 15:51.58 | ntz | WIMPy: http://susepaste.org/view/raw/97574676 |
| 15:52.07 | ntz | this is exact dialplan (extensions.ael) |
| 15:52.34 | WIMPy | Oh. AEL. |
| 15:52.47 | ntz | I was playing with Set(CALLERID(foo)=xyz), that's why it's commented out |
| 15:53.27 | WIMPy | Normal caller ID is in CALLERID(num). |
| 15:53.59 | ntz | WIMPy: I'd start first with checking from sip debug output if sending side sends it |
| 15:54.36 | WIMPy | The sending t38modem does not send any caller ID. |
| 15:55.44 | ntz | WIMPy: http://susepaste.org/view/raw/15215408 |
| 15:56.00 | ntz | here is some sort of correct from and to destinations |
| 15:56.58 | WIMPy | The only from part I see there is the username. |
| 15:57.02 | ntz | hmm |
| 15:57.28 | ntz | can I somehow pair on the astrisk side that 10.0.255.211 has a prefix 111211 ? |
| 15:58.07 | WIMPy | You can set caller ID in a peer, yes. |
| 16:01.11 | ntz | but I can't do it on the hylafax side .... |
| 16:01.17 | ntz | :( |
| 16:01.52 | ntz | can I somehow set on asterisk side that call from 10.0.255.211 will have 111211XXX ? |
| 16:01.58 | ntz | as the CID |
| 16:04.18 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1::1:3) |
| 16:06.17 | WIMPy | You can set caller ID in a peer, yes. peer as in sip.conf. |
| 16:11.02 | ntz | ok |
| 16:14.25 | ntz | WIMPy: okay, CID works now !!!! good |
| 16:14.34 | ntz | now I have to make working DID |
| 16:18.36 | WIMPy | I still have no idea what you mean there. |
| 16:24.50 | ntz | that f****** DID does not work .... I am not able to see on the receiving side the number that is supposed to be receiving a fax |
| 16:28.34 | WIMPy | Ah, the called number. |
| 16:29.17 | WIMPy | Then you should try to Dial(sip/theModemsPeerName/TheCalledNumber)/ |
| 16:29.19 | WIMPy | . |
| 16:30.36 | ntz | superb, thanks |
| 16:32.53 | ntz | WIMPy: I guess right in here, eg Dial() part from my extensions.ael: Dial(SIP/${EXTEN}@T38modem_mgec211,15,g); |
| 16:33.41 | WIMPy | yes |
| 16:34.12 | WIMPy | But you already have it there, just with the @-syntax. |
| 16:34.29 | WIMPy | So maybe your EXTEN is no longer valid. |
| 16:35.19 | WIMPy | No, looks good. |
| 16:35.46 | ntz | hmm |
| 16:35.50 | WIMPy | BTW: Thanks for using raw text pastebins. |
| 16:36.11 | ntz | WIMPy: seriously or ironic ? |
| 16:36.26 | WIMPy | Seriousely. |
| 16:36.36 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 16:36.37 | ntz | good :), I also like it, you're welcome |
| 16:36.43 | WIMPy | It's quicker and more readable. |
| 16:38.43 | ntz | I'd be so lucky if I'd be able to see on receiving side a DID :( .... isn't there a way that I modify somehow my Dial() command ? |
| 16:39.25 | WIMPy | That must be a t38modem thing. |
| 16:39.39 | ntz | hmm |
| 16:40.19 | WIMPy | Anyway I'm away for the evening NOW. |
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| 18:42.51 | ||cw | ntz: that's typically done by modifying the CID with the DID info, like not sending the name, bit the CID number field has the CID, and the CID name field has the DID, but the provider needs to do this |
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| 23:08.51 | scv | fun times |
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| 23:09.05 | scv | grandstream is using freepbx code in their ucm61xx without acknowledging it |
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