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| 00:28.46 | klow | Lost my connection for a bit.. anyone around see my last few posts ~1 ago re: TLS connections? |
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| 01:11.59 | Get_The_Fish | Can anyone tell me how to set the default outbound endpoint? I have the option set pjsip.conf but it just doesnt see to want to take it. |
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| 03:38.55 | socomm | I recently got a cisco 7961 phone to use with Asterisk. I'm able to program the first button for my SIP user, but I'm wondering if I can use the other 5 buttons for the same user. |
| 03:39.20 | socomm | I tried adding the SIP user to button 2 but there was a conflict. |
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| 04:15.24 | MRH2 | hey anyone know hardware phones that can be also connected via USB to a PC to work as Mic and speaker as well (switching between the two as needed)? |
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| 04:17.36 | MRH2 | sort of thing person has normal calls then needs to join a webex on the PC and use the same headset/handset/speaker by pressing a button |
| 04:18.20 | MRH2 | flipping between SIP Connection and PC connection |
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| 04:25.31 | drmessano | socomm: Cant register 5 times using the same credentials |
| 04:25.52 | drmessano | socomm: Maybe you can set up SLA or BLF or one of those other awful things |
| 04:26.18 | socomm | drmessano: Figured as such. Thanks for the pointer |
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| 05:09.00 | [TK]D-Fender | <drmessano> socomm: Cant register 5 times using the same credentials <- can with PJSIP |
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| 12:23.54 | stefan27 | When i use AMI action "SIPnotify" asterisk creates an outgoing NOTIFY sip message where the contact header is Contact: <sip:asterisk@127.0.0.1:8060> but usually when asterisk sends NOTIFYs it populates it with its (local) ip : Contact: <sip:asterisk@192.168.1.227:8060> |
| 12:24.05 | stefan27 | anyone knows why? |
| 12:41.09 | stefan27 | apparently if I add "udpbindaddr=192.168.1.227:8060" instead of 0.0.0.0:8060 to sip.conf i get the right ip in the contact header even in the AMI-initiated SIPnotify, but that's not really what I wanna do |
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| 13:06.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:07.55 | [TK]D-Fender | stefan27: use the IP of the interface it uses for the packet unless you proper define your subnets and give it an external IP it NAT'd |
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| 13:20.35 | onebree | Good morning, and happy Friday |
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| 13:25.55 | onebree | In sip.conf, we have "dtlssetup=somepass", but the only valid options are active, passive, and auto. |
| 13:25.55 | onebree | At one point did dtlssetup accept a password as a value? |
| 13:27.32 | stefan27 | [TK]D-Fender, yeah thx I can do that. I also asked the -dev channel why cli_sip_notify and manager_sipnotify behave differently w.r.t. the own-ip-lookup |
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| 13:28.42 | [TK]D-Fender | stefan27: I'd like to see samples showing the difference |
| 13:28.48 | stefan27 | onebree isn't it active, passive or actpass? |
| 13:29.11 | Lope | musiconhold.conf [default]; mode=files; directory=/mymusic; random=yes; |
| 13:29.22 | Lope | <PROTECTED> |
| 13:29.45 | Lope | My music files are mp3's. I've got format_mp3 loaded |
| 13:30.33 | Lope | However my mp3s are not some special bitrate and sample freq...? |
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| 13:31.32 | onebree | stefan27: You are correct -- actpass instead. Either way, our config has a password, while the config for 13 only has 3 enum options |
| 13:33.03 | stefan27 | probably just a typo then, the field dtlssetup should never contain a password, just one of those 3 strings? |
| 13:33.55 | onebree | IDK, I do not think it was a typo -- by "somepass" I do not mean "actpass" but rather "s0m3*pa55" |
| 13:34.00 | Lope | I also tried mode=mp3 |
| 13:37.20 | [TK]D-Fender | Lope: PASTEBIN everything. Full config file, class dump from CLI, folder contents with "ls -la THEFOLDER" along with the command as issued |
| 13:38.10 | Lope | TKD: thanks, what is class dump? |
| 13:40.23 | onebree | stefan27: Do you think I should do "dtlssetup=actpass"? It seems like the best choice of the three, and was suggested in the sipml5 tutorial |
| 13:41.17 | [TK]D-Fender | Lope: just the whole files |
| 13:41.57 | stefan27 | onebree, yes that's what we use |
| 13:42.29 | onebree | Okay, thank you. |
| 13:48.07 | Lope | TKD: the dir and the conf file http://codepad.org/Xmx3yuG6 |
| 13:54.58 | [TK]D-Fender | mp3 = wrong |
| 13:55.09 | [TK]D-Fender | that should be "files", and why are the files themselves owned by root? |
| 13:55.24 | [TK]D-Fender | You shouldn't be running * as that either |
| 13:57.34 | wdoekes | (nothing wrong with files being owned by root if asterisk is not supposed to write to them) |
| 13:58.37 | Lope | TKD: I'm not running asterisk as root. I use service asterisk start/stop/restart |
| 13:58.54 | Lope | I thought what wdoekes said. |
| 13:59.12 | Lope | TKD: when mode was set to files it didn't work either. |
| 13:59.32 | Lope | When running core reload with debug and verbose 10 it doesn't mention musiconhold.conf or mp3s etc. |
| 13:59.46 | Lope | No complaints or acknowledgements about any of it |
| 13:59.58 | Lope | hmm, maybe I'm missing a module. |
| 14:00.29 | Lope | ah I think I was missing res_musiconhold.so |
| 14:03.33 | Lope | WARNING[2754][C-00000000]: mp3/interface.c:217 decodeMP3: Junk at the beginning of frame 49443303 |
| 14:04.14 | Lope | Getting somewhere. I don't know which mp3 it has a problem with though. it didn't say. Oh cool it's playing :) it was just a warning. |
| 14:05.08 | Lope | so mode=mp3 is supposed to be loud? |
| 14:06.32 | Lope | it doesn't like it when I say mode=mp3 . WARNING[2755][C-00000001]: res_musiconhold.c:884 _get_mohbyname: Music on Hold class 'default' not found in memory. Verify your configuration. |
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| 14:07.26 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:08.39 | Lope | can I pass multiple options to mode? |
| 14:08.45 | Lope | like mode=files,mp3 > |
| 14:08.51 | Lope | like mode=files,mp3 ? |
| 14:10.14 | Lope | WARNING[2753]: res_musiconhold.c:1290 _moh_register: Don't know how to do a mode 'files|mp3' music on hold |
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| 14:12.57 | [TK]D-Fender | <PROTECTED> |
| 14:13.01 | [TK]D-Fender | Stop using "mp" |
| 14:13.05 | [TK]D-Fender | thta uses an EXTERNAL player |
| 14:13.37 | [TK]D-Fender | And make sure you have format_mp3 compiled in |
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| 14:36.09 | MrMeek | I've got some old GrandStream GXPs phones behind nat, connecting to remote asterisk... they all work for a while after register than randomly become UNREACHABLE o.o |
| 14:36.39 | MrMeek | any heard of nat issues with grand streams, or any other ideas? shorter qualify, stun? |
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| 14:49.02 | Lope | TKD ok thanks :) |
| 14:52.21 | Lope | BTW I realized using mp3s is kind of pointless. When you convert the mp3 to a signedint 8kHz file it's basically the same size as the mp3, without the CPU overhead. There are much bigger issues though. For example I needed to cut the beginning of my music file off so that the caller doesn't hear the whole quiet buildup, they just hear the medium buildup. And then also I had a problem where the voip call was clicking and popping due to the bass (i tried redu |
| 14:52.21 | Lope | cing the volume to -9dB and it still clicked and popped on every bass beat. So then I ran a highpass filter of 800Hz on it and it worked even at 0dB (full volume). I actually settled on 12dB because VoIP music sounds like crap anyway, nobody wants to hear it blasting at them. It's basically there to keep you ever so slightly entertained until somebody picks up. |
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| 15:09.54 | Lope | what formats should I convert my music on hold to? I've got wav and gsm |
| 15:10.33 | [TK]D-Fender | The the one your calls use |
| 15:11.10 | Lope | I see g722 is common in the apt repos. My calls will likely use ulaw but I support all the common stuff |
| 15:11.51 | [TK]D-Fender | then just use what your calls actually use |
| 15:16.29 | onebree | Lope: If you still want to support the "default" codecs, then put them last in your list. You can accept multiple codecs "allow=ulaw,g722" in a comma-separated list, with the first being top priority/preference |
| 15:18.20 | Lope | onebree: thanks, yes I've done that :) |
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| 15:21.04 | Lope | I don't know how to test my different format MOH files other than to take all out of the dir except one, then test, then repeat with each time. |
| 15:21.09 | Lope | each file. |
| 15:23.12 | Lope | well, they all work. testing done. |
| 15:24.48 | Lope | What's a good miniumum frequency to filter out for MOH? I read 300Hz somewhere but when I tried 500Hz it was still clicking and popping, even 700Hz IIRC. 800Hz resolved the clicking and popping on the "bass" issue. |
| 15:25.31 | Lope | Bad english. What's a good high pass frequency to use? |
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| 15:32.09 | Lope | okay with a gain of -12 a high pass of 300Hz actually works without pops and clicks. |
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| 15:39.41 | Lope | What's the command to reload extensions.conf? |
| 15:40.16 | [TK]D-Fender | dialplan reload |
| 15:40.22 | Lope | Thanks |
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| 17:35.28 | cusco | hi |
| 17:35.36 | cusco | can I use a register string with a escaped password? |
| 17:35.52 | cusco | password from voip provider has a ? interrogation mark |
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| 18:35.23 | alsaiduq | hello |
| 18:35.35 | alsaiduq | how does cdr records work? |
| 18:36.10 | alsaiduq | actually i dont exactly what CDR is supposed to be |
| 18:36.21 | alsaiduq | i dont know exactly* |
| 18:40.06 | [TK]D-Fender | Yes, they work |
| 18:40.31 | [TK]D-Fender | There is a chapter on this in the book. |
| 18:40.33 | [TK]D-Fender | ~book |
| 18:40.33 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:40.35 | [TK]D-Fender | ^^^ |
| 18:43.15 | onebree | alsaiduq: CDR means call detail records. Basically, in/outgoing numbers, duration of calls, etc. |
| 18:46.53 | alsaiduq | onebree: but where does CDR stores these registers? |
| 18:47.12 | [TK]D-Fender | Wherever you configured it to in cdr.conf |
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| 19:07.48 | onebree | Is there such a think as "sips"? The URI of this pjsip contact is: sips:286@example.com:8089, but I have it configured in pjsip.conf as just sip (without the S) |
| 19:08.40 | Kunsi | It is |
| 19:08.50 | dadrc | sips is SIP over TLS |
| 19:08.54 | Kunsi | It means sip over tls (port 5061) |
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| 19:20.12 | onebree | Thank you, both! |
| 19:20.50 | onebree | Any chance someone knows that this error message means? I have tried a lot but still get it: |
| 19:20.51 | onebree | SRTP unprotect failed with: authentication failure 10 |
| 19:21.11 | onebree | (Sometimes it's error code 110, there it no pattern to which shows up when) |
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| 19:52.05 | klow | Question. Clients connecting to an Asterisk via TLS - they should maintain a *persistent* connection to the Asterisk server, server TLS port 5061 , correct? And when a call is destined for an endpoint, it would receive an INVITE through the TLS connection which the endpoint initiated to the server and maintains .. is this correct? |
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| 19:53.18 | klow | Basically I am wondering if Asterisks would be trying to connect TO endpoints with TLS for signaling , or if it would just communicate to endpoints via the existing TLS connection under normal circumstances |
| 19:57.19 | WIMPy | I think it's both possible. |
| 19:58.07 | WIMPy | But even if you use TCP without TLS, it would usually be only the initial connection. |
| 19:58.46 | klow | I see. The reason I ask is I am seeing calls fail with TLS connect failure, but the phones are reigistered with TLS fine |
| 19:58.52 | Chainsaw | The thing about TLS is that the initial connection will always be unencrypted. It is "upgraded" from that state with STARTTLS. |
| 19:59.00 | klow | so Im wondering why Asterisk would say there is a TLS connection failure .. the phone is already registered. |
| 19:59.23 | klow | when I say I see that I mean in the sip debug log |
| 20:00.21 | WIMPy | Huh? If it uses STARTTLS, why do we have an extra port, then? |
| 20:02.07 | Chainsaw | WIMPy: That's a point. I'm so used to seeing TLS with STARTTLS. |
| 20:02.16 | Chainsaw | klow: Forget what I said. It was wrong. |
| 20:02.24 | Chainsaw | klow: Go with WIMPy on this one. |
| 20:03.07 | WIMPy | I have never looked in to it. It just wouldn't make any sense. ... Which doesn't mean anything, off course. |
| 20:05.14 | Chainsaw | WIMPy: 5061 vs 5060 means you are obviously correct. Which is the best kind of correct. |
| 20:05.55 | WIMPy | It *seems* obvious. But I don't take anything for granted here. |
| 20:08.14 | klow | Hmm |
| 20:08.30 | klow | So I changed my transport in pjsip and my soft phones to TCP in case I need to more easily wireshark this |
| 20:08.43 | klow | basically i have 2 soft phones, they register, but don't ring when I call each other |
| 20:08.50 | klow | I'm seeing this in the log |
| 20:08.53 | klow | [Jan 29 12:05:54] DEBUG[7705] res_pjsip_session.c: Method is INVITE |
| 20:08.53 | klow | [Jan 29 12:05:54] DEBUG[7705] pjsip: tcpc0x7f323001 ...TCP client transport created |
| 20:08.53 | klow | [Jan 29 12:05:54] DEBUG[7705] pjsip: tcpc0x7f323001 ...TCP transport 10.50.55.10:46201 is connecting to 73.53.31.109:57652... |
| 20:08.53 | klow | [Jan 29 12:05:54] DEBUG[7705] res_pjsip_multihomed.c: Re-wrote Contact URI host/port to 10.50.55.10:46201 |
| 20:09.07 | klow | then the call fails with "no answer" |
| 20:09.30 | klow | so that actual says that the server (the nat ip) is connecting TO the endpoint .. the public IP |
| 20:10.21 | WIMPy | So that's already at least two seperate issues. |
| 20:12.28 | klow | ya? So my server has the proper port forwarding configured (its at a data center) and that public IP is my home public IP, also behind my firewall but should open with uPnP or whatever. Works fine with my other PBX (freepbx/chan_sip) |
| 20:12.35 | klow | but configuring this one from the ground up with pjsip/asterisk |
| 20:14.09 | *** join/#asterisk bmg505 (~leon@196-210-48-11.dynamic.isadsl.co.za) |
| 20:19.43 | *** join/#asterisk xnaron (~xnaron@S0106b4750e5de3b2.ed.shawcable.net) |
| 20:23.23 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 20:29.13 | alsaiduq | quit |
| 20:30.56 | klow | So when you type "pjsip show endpoints" in asterisk CLI |
| 20:31.06 | klow | it shows you one or more of these for each endpoint: |
| 20:31.06 | klow | Contact: 6000/sip:6000@100.113.39.222:59194;transpo 4adde25cdd Unknown nan |
| 20:31.35 | klow | I assume that means that 100.113.39.222 should be reachable via port 59194 from the server |
| 20:32.06 | klow | as in, this contact is a listing for how the asterisk server can reach the phone BACK for SIP signaling (as opposed to RTP?) |
| 20:33.15 | [TK]D-Fender | Correct |
| 20:33.23 | *** join/#asterisk bmg505 (~leon@196-215-6-13.dynamic.isadsl.co.za) |
| 20:33.24 | [TK]D-Fender | There is no state or setting for RTP. |
| 20:33.37 | [TK]D-Fender | That is negotiated in the call. |
| 20:42.21 | *** join/#asterisk alsaiduq (~emerson@200.239.65.110) |
| 20:42.25 | alsaiduq | hell |
| 20:42.27 | alsaiduq | hello |
| 20:43.40 | alsaiduq | do i can't connect throught IAX asterisk 1.8 and asterisk 11.7? |
| 20:45.08 | *** join/#asterisk Echo6 (~Echo6@64.136.247.50) |
| 20:50.40 | klow | thanks [TK]D-Fender . |
| 20:51.11 | klow | next question - NAT traversal. A phone, or soft phone etc, would normally use what , uPnP to tell it's firewall to let the Asterisk server connect back to it? |
| 20:51.49 | klow | Using the same soft phones with FreePBX I don't seem to be having these NAT problems - but my ground-up configuration is. Does the server TELL the client to try uPNP or some other auto-port forwarding mechanism? |
| 20:53.13 | klow | the tcp ports (when using tcp transport) are not open, I cannot telnet to them from my asterisk machine as listed in pjsip show endpoints . but I'm not sure why they aren't opening, the client must initiate a NAT firewall asking it to open the ports I assume (no way to statically configure my scenarios I'm afraid I have to do everything possible for NAT traversal configuration) |
| 20:54.34 | *** join/#asterisk bmg505 (~leon@196-210-200-143.dynamic.isadsl.co.za) |
| 20:56.48 | alsaiduq | [TK]D-Fender: do i can't connect throught IAX asterisk 1.8 and asterisk 11.7? |
| 20:59.58 | [TK]D-Fender | That sentence makes no sense\ |
| 21:04.36 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 21:04.40 | alsaiduq | no problem, solved it already |
| 21:04.48 | alsaiduq | but now i'm getting this |
| 21:05.07 | alsaiduq | No IAX provisioning configuration found, IAX provisioning disabled. |
| 21:09.48 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 21:11.04 | alsaiduq | [TK]D-Fender: No IAX provisioning configuration found, IAX provisioning disabled. Do you know what that means? |
| 21:11.39 | [TK]D-Fender | It means you are using IAX for provisioning. |
| 21:12.00 | [TK]D-Fender | Which NOBODY uses because that was only for the IAXy ATA which has been dead for 10 years |
| 21:12.07 | [TK]D-Fender | It means nothing |
| 21:17.40 | *** join/#asterisk bmg505 (~leon@196-210-200-237.dynamic.isadsl.co.za) |
| 21:27.06 | alsaiduq | its strange.. |
| 21:27.19 | alsaiduq | can't get why its happening... |
| 21:27.40 | alsaiduq | the asterisk which is giving that error is 11.7 version |
| 21:28.16 | alsaiduq | the other one, version 1.8, can conect to 11.7 seemlessly |
| 21:29.32 | alsaiduq | how to stop asterisk to use IAX for provisioning? |
| 21:30.30 | alsaiduq | i guess thats not the problem |
| 21:32.00 | alsaiduq | seemingly asterisk 11.7 connected to 1.8 but when i type "iax2 show peers" it outputs "argentina (null) (D) 255.255.255.255 0 UNKNOWN |
| 21:32.05 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 21:37.29 | alsaiduq | perhaps things will be solved on monday |
| 21:37.59 | alsaiduq | gotta go now |
| 21:38.01 | alsaiduq | bye |
| 21:38.36 | *** join/#asterisk bmg505 (~leon@196-210-200-131.dynamic.isadsl.co.za) |
| 21:45.41 | *** part/#asterisk onebree (d1bf0df7@gateway/web/cgi-irc/kiwiirc.com/ip.209.191.13.247) |
| 21:55.21 | *** join/#asterisk jhester (jhester@nat/digium/x-whodzypxejshnxof) |
| 21:59.16 | *** join/#asterisk bmg505 (~leon@196-210-200-131.dynamic.isadsl.co.za) |
| 22:00.31 | *** join/#asterisk tzafrir (~tzafrir@91.183.149.90) |
| 22:05.54 | *** join/#asterisk italorossi (~Adium@179.156.47.253) |
| 22:17.13 | *** join/#asterisk C1ph3r4 (~C1ph3r@187.58.32.162) |
| 22:17.55 | *** join/#asterisk bmg505 (~leon@196-210-200-237.dynamic.isadsl.co.za) |
| 22:20.20 | *** join/#asterisk captain118 (~Adium@21.sub-70-193-103.myvzw.com) |
| 22:34.24 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
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| 22:56.31 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 23:02.00 | *** join/#asterisk bmg505 (~leon@196-210-48-89.dynamic.isadsl.co.za) |
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| 23:32.59 | klow | If remote endpoints are behind a NAT, what mechanism would they usually use to ensure that the SIP signaling port is open to accept connections from the Asterisk server about a call coming ? |
| 23:34.59 | klow | my soft phones are registering fine, and I see their contact IP/port in pjsip show endpoints .. but they are not reachable via those ports - I'm not sure how those ports would get opened, via uPnP ideally? The ports seem to by dynamically allocated, at least on this soft phone .. so it isn't a matter of a manual port forward settings - plus the phone may roam to any number of different NATed environments as it is a mobile app . |
| 23:44.40 | *** join/#asterisk bmg505 (~leon@196-215-18-18.dynamic.isadsl.co.za) |
| 23:56.46 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 23:57.17 | *** join/#asterisk jameswf_ (uid27319@gateway/web/irccloud.com/x-izyitfqiajjwdmmz) |
| 23:57.50 | *** part/#asterisk kharwell (kharwell@nat/digium/x-yidtxarktbzkwvqv) |