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| 01:14.16 | snadge | how do you set caller id with IAX? |
| 01:14.42 | snadge | our network requries p-asserted-identity to pass callerid upstream |
| 01:15.01 | snadge | but instead the remote party id is being set |
| 01:16.17 | snadge | ... tell the customer not to use iax.. :p |
| 01:20.16 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:20.16 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 01:22.12 | snadge | yeah.. so we would have to convert the remote party id to a peer asserted id.. before we send it upstream |
| 01:22.25 | snadge | im not sure i can be bothered doing that |
| 01:22.33 | snadge | no excuse for using IAX these days anyway |
| 01:22.36 | snadge | its lame |
| 01:23.01 | Samot | So you're user is connecting via IAX and you're sending it out SIP? |
| 01:24.15 | Samot | And you're upstream is rejecting the call you're sending? |
| 01:24.39 | Samot | s/you're/your/ |
| 01:26.32 | snadge | no.. its not rejecting.. its just not got the caller id that they wanted |
| 01:26.43 | Samot | OK |
| 01:26.50 | Samot | How are you sending it? |
| 01:26.55 | snadge | and the simple solution is.. switch to using sip, and set p-asserted-identity ;) |
| 01:27.10 | Samot | You're customer isn't sending it wrong. |
| 01:27.12 | snadge | via sip.. to our upstream provider.. with a remote-party-id header, thats ignored |
| 01:27.35 | Samot | You're trunk with the SIP provider is going to state how you send the caller ID format. |
| 01:27.53 | Samot | I don't even think IAX supports RPID. |
| 01:28.01 | Samot | I think it's all standard FROM header. |
| 01:28.16 | snadge | iax is all binary.. theres no from header as such |
| 01:28.28 | snadge | but whatever the equivalent of that is .. sure |
| 01:29.14 | Samot | What is the rpid setting in the trunk to the provider? |
| 01:29.55 | Samot | sendrpid |
| 01:31.38 | snadge | not specified apparently |
| 01:32.03 | snadge | so maybe i could just put.. sendrpid=pai in the trunk setting |
| 01:32.20 | Samot | Yes, that will send all the callerid in PAI format. |
| 01:32.30 | snadge | what version of asterisk is required for this option? |
| 01:32.30 | Samot | Yes, that will send all the callerids in PAI format. |
| 01:32.34 | Samot | 1.8 |
| 01:32.37 | snadge | rofl |
| 01:32.43 | Samot | So like... |
| 01:32.55 | snadge | so umm.. we still have 1.6 on our main core systems |
| 01:33.18 | snadge | talk about legacy |
| 01:33.20 | Samot | Well that's... |
| 01:33.24 | Samot | Awful. |
| 01:33.34 | snadge | one of our servers runs asterisk 11 though |
| 01:33.47 | snadge | and our house of cards is patched to work with that system.. so i might just tell customer to use that one |
| 01:34.18 | snadge | the whole platform is going to be scrapped.. and rolled into the parent companies systems within the next year or two |
| 01:34.24 | snadge | so its just kicking the can down the road until then |
| 01:34.49 | snadge | they've inherited a legacy pile of unmaintained crap.. (of course).. with thousands of paying customers who would be horrified if they knew |
| 01:35.45 | snadge | but to their credit.. when something works.. and you leave it alone.. thats how you keep customers |
| 01:36.06 | snadge | constantly upgrading and changing stuff, just because its "better".. is a good way to lose them, and create more work in the process ;) |
| 01:36.41 | Samot | Or you know.. |
| 01:36.50 | Samot | Being able to use a standardized callerid format. |
| 01:37.10 | Samot | Having numerous bugs fixed. |
| 01:37.10 | snadge | fortunately most customers don't use IAX.. but theres a few that still do.. god only knows why |
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| 01:40.50 | Samot | Well.. |
| 01:41.02 | Samot | IAX on a 1.6 box, that means some of these customers could be at least 10 years old. |
| 01:41.38 | Samot | Or did it because of poor bandwidth or networking, at least one of which has had to improve in the last decade. |
| 01:42.17 | Samot | But it also could be they are still using IAX because no one told them they could use SIP... |
| 01:42.34 | *** join/#asterisk Chotaire (chotaire@oahu.chotaire.net) |
| 01:42.46 | snadge | yeah.. im about to tell this guy he can use sip |
| 01:42.46 | Samot | That's a part of the "it's running and working so don't touch" logic. |
| 01:42.59 | snadge | and thats exactly the angle hes using to complain about it |
| 01:43.06 | snadge | "this has been working for years.. i dont want to touch it.. fix it" |
| 01:43.14 | Samot | Yeah |
| 01:43.23 | Samot | Have had those conversations. |
| 01:43.34 | Samot | Then you say "But you can use this and here's why it's better...." |
| 01:43.55 | Samot | Then they'll most likely say "Oh well I didn't know that...why didn't someone tell me this was better and get me on it?" |
| 01:47.09 | snadge | i guess so.. assuming you get someone rational and not stupid |
| 01:47.17 | snadge | which is a pretty big assumption really |
| 01:47.18 | snadge | :P |
| 01:47.21 | Samot | There are benefits to keeping your infrastructure up-to-date. Not all of it is customer facing. |
| 01:47.37 | snadge | yeah totally.. our parent organisation are fully aware of that |
| 01:47.42 | snadge | and update daily from git |
| 01:47.43 | Samot | Not saying you have to be bleeding edge and update with every release.. |
| 01:47.46 | snadge | j/k |
| 01:48.01 | Samot | But a solid upgrade and maintenance plan stops this. |
| 01:48.12 | snadge | they at least update though.. yeah.. and sometimes stuff happens.. thats just the nature of the beast |
| 01:48.39 | snadge | how you communicate that to your customers.. and how you learn from it to improve your processes is something else |
| 01:48.42 | Samot | Not every update/upgrade goes smooth. |
| 01:48.57 | Samot | That's what rollback part of the plan is for. |
| 01:49.16 | Samot | I use maintenance notices. |
| 01:49.26 | Samot | And have routine windows. |
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| 02:30.36 | sarthor | Hi, I have Dlink ip phone DPH400SE, I want connect this IP phone with my Openvpn server, while it support Openvpn, but I am unable to find any manual, how to upload openvpn config files. I clicked on brows, and tried to upload, but says error, please let me know if some one knows, how to configure IP Phone for Openvpn. Thanks. |
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| 05:40.19 | *** join/#asterisk Haris (~haris@unaffiliated/haris) |
| 05:40.21 | Haris | hello all |
| 05:40.37 | Haris | the https://www.voip-info.org/wiki/index.php?page_id=4194&comments_page=1 page needs to be updated to make the syntax 'queue show' in contrast to 'show queue' |
| 05:40.49 | Haris | I was stuck with show queue for a bit |
| 05:40.54 | Haris | not working that is |
| 05:41.29 | Haris | https://www.voip-info.org/wiki/view/asterisk+cli+command+queue+show <- this page has it right |
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| 05:58.32 | igcewieling | try the asterisk wiki |
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| 06:17.45 | drmessano | voip-info is fucking useless |
| 06:17.52 | Haris | If I have an asterisk server or SIP exchange |
| 06:18.07 | Haris | how do I control how many simultaneous calls can a customer run through me ? |
| 06:18.35 | Haris | information on voip-info needs to be organized or presented a bit better |
| 06:18.49 | drmessano | So go fix it |
| 06:18.53 | Haris | how ? |
| 06:18.58 | drmessano | Sign up for an account |
| 06:19.01 | drmessano | Make changes |
| 06:24.14 | Haris | ah |
| 06:24.20 | Haris | has 1 calls (max unlimited) in 'ringall' strategy (49s holdtime), W:0, C:210, A:201, SL:0.0% within 0s <--- what's the holdtime ? |
| 06:26.13 | Haris | my server with 5 agents is .. like .. totally idle |
| 06:26.27 | Haris | <PROTECTED> |
| 06:34.45 | Samot | What are you trying to exactly figure out? |
| 06:47.26 | drmessano | Life |
| 06:47.54 | Samot | Like he pastes a 1.4 example from voip-info, not saying that's what it is... |
| 06:48.06 | Samot | Then talks about *his* agents and queues |
| 06:48.31 | Samot | So for a second there, I thought that was *his* queue's stats. |
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| 07:14.25 | Haris | the url I pasted has the 'show queue' command in it |
| 07:14.42 | Haris | this might have been true/working in a much older version of asterisk |
| 07:14.49 | Haris | on 11+ its 'queue show' |
| 07:15.06 | [TK]D-Fender | <Haris> the https://www.voip-info.org/wiki/index.php?page_id=4194&comments_page=1 page needs to be updated to make the syntax 'queue show' in contrast to 'show queue' <- This WIK is decrepic junk |
| 07:15.09 | Haris | that's the part that needs updated. this is for the URL |
| 07:15.29 | [TK]D-Fender | and should not be used as anything in the realm of first 5 places to look... |
| 07:15.31 | Haris | it shows up in alot of google searches |
| 07:15.43 | [TK]D-Fender | That doesn't mean much |
| 07:16.04 | [TK]D-Fender | use the official wiki, then sample configs, docs in the tarball, etc all before thinking of using that site |
| 07:16.12 | [TK]D-Fender | And keep in mind 11 = disco'd |
| 07:16.44 | Haris | hmm |
| 07:17.08 | Samot | show queue hasn't worked since 1.4 or something like that |
| 07:17.13 | Haris | hmm |
| 07:17.18 | Samot | Even that web page covers that. |
| 07:17.30 | Samot | queues show |
| 07:17.46 | Samot | You still haven't answered what you are exactly trying to figure out... |
| 07:18.25 | [TK]D-Fender | <Haris> has 1 calls (max unlimited) in 'ringall' strategy (49s holdtime), W:0, C:210, A:201, SL:0.0% within 0s <--- what's the holdtime ? <- My guess .. 49 seconds |
| 07:18.36 | Haris | I mean what does holdtime mean in this |
| 07:18.47 | [TK]D-Fender | how long you're on hold. |
| 07:18.48 | Haris | is it a setting or an average |
| 07:19.01 | Samot | It's the_current_ average. |
| 07:19.02 | [TK]D-Fender | it's a STATUS page, not a configuration dump |
| 07:19.14 | Haris | good point |
| 07:19.35 | [TK]D-Fender | recent average hold time before being answered. |
| 07:19.42 | [TK]D-Fender | it's exactly as obvious a thing as it seems |
| 07:19.48 | Haris | =) |
| 07:20.15 | [TK]D-Fender | <Haris> how do I control how many simultaneous calls can a customer run through me ? <- it's your dialplan. group_count your channels, log in a DB, or whatever other method you feel like. It's all on you |
| 07:20.41 | Haris | dial plan > group_count. thank you! |
| 07:22.41 | Samot | group_count alone isn't the answer. |
| 07:23.20 | Haris | I have agent(s) who want to go on break. maybe lunch. maybe prayers. maybe tea, etc etc. for any given reason. right now I make them turn off their soft phones to .. in effect log-out. Is there a way to press some button or setting on the sip client to make the agent not receive calls during the time they are .. away ? |
| 07:23.39 | Samot | Yes. |
| 07:26.40 | Samot | The members can pause themselves, with a reason. |
| 07:29.37 | Samot | Also, if this is one of your FreePBX machines then you are fighting a whole different battle. |
| 07:32.07 | drmessano | How dare you accuse him of running FreePBX |
| 07:32.14 | drmessano | How dare you, SIR |
| 07:32.20 | Samot | Because he is. |
| 07:32.29 | drmessano | FOR SHAME, SIR |
| 07:32.54 | drmessano | Do you know who else accused people of running FreePBX? |
| 07:32.57 | drmessano | HITLER |
| 07:33.07 | Samot | Lol. |
| 07:33.14 | Samot | Haris = German. |
| 07:34.02 | drmessano | Alex Jones would be pleased of the irony |
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| 07:34.31 | Samot | Under the floor board. |
| 07:34.33 | Haris | hmm |
| 07:34.43 | Haris | pause ? |
| 07:34.46 | Haris | hmm |
| 07:34.48 | Samot | Yes, pause. |
| 07:36.29 | Samot | Even voip-info has links on their queue pages to the other queue applications/functions. |
| 07:36.43 | Samot | Like PauseQueueMember... |
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| 07:42.26 | Haris | hmm |
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| 07:45.38 | Haris | when a Q member is on pause, calls coming in a Q won't go to them. Cool |
| 07:46.30 | Haris | rather than .. on numbered button press .. is there another way to set pause for a Q member, when they click pause on their soft phone ? |
| 07:49.16 | Haris | Q 3) We don't hear any .. sort of bell .. for outgoing calls .. on our asterisk box. Its just empty time/space when an outgoing call is in dialing state. Can we change that on our box ? |
| 07:50.07 | Haris | Q 4) SL: came down from 100 % to 96.4! |
| 07:50.32 | Haris | has 0 calls (max unlimited) in 'rrmemory' strategy (15s holdtime, 136s talktime), W:0, C:28, A:5, SL:96.4% within 60s |
| 07:52.05 | Haris | reading on it |
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| 08:03.31 | tomcruise | Having this strange issue with 14.3.0 and ConfBridge. When recording a participants name (x=0, open writing: /var/spool/asterisk/confbridge/confbridge-name-8888-1489065962.94 format: sln, 0x7f67c0011058), the file does get saved (as .sln). Playback to other participants however is just silence. I noticed the playback tries to open the same filename but then with extension .slin instead of .sln. Is |
| 08:03.37 | tomcruise | this a known issue or config setting maybe? |
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| 08:38.49 | sotoz | I'm trying to use the rfc2833 dtmf_mode with pjsip but I get the "ERROR[17236]: config_options.c:738 aco_process_var: Error parsing dtmf_mode=rfc2833 at line 46" message. If I switch to rfc4733 it works but I'm not sure if the operators that I'm using support rfc4733. They say they only support rfc2833 and I'm not sure if those two are fully compatible. I know that 4733 obsoleted 2833 but what that means in terms of compatibility? |
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| 08:42.24 | tomcruise | Should work fine sotoz |
| 08:42.34 | tomcruise | Set your PBX to 4733 |
| 08:43.22 | tomcruise | They are pretty much identical. |
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| 08:43.43 | sotoz | ok, thank you. Is there anything that could go wrong that you can think of? |
| 08:43.55 | tomcruise | No, never encountered any issues. |
| 08:44.24 | sotoz | ok then :) thanks |
| 08:44.33 | tomcruise | np :) |
| 08:44.54 | tomcruise | you're dutch as well i see, which sip provider are you using? |
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| 08:54.41 | sotoz | tomcruise nexmo atm |
| 08:54.58 | tomcruise | ok |
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| 10:11.09 | wasanzy | hello |
| 10:11.25 | sotoz | hi |
| 10:12.13 | wasanzy | I have configured sipML5, I can login without https but with https, I am getting "Disconnected: Failed to connet to the server" |
| 10:12.26 | wasanzy | nothing has been indicated in the logs |
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| 10:18.18 | dan_j | wasanzy: firewall? |
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| 10:27.49 | MacroMan | wasanzy, Are you using a valid cert? |
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| 10:45.07 | *** join/#asterisk wasanzy (~wasanzy@41.66.254.58) |
| 10:45.34 | wasanzy | I was able to connect using http. now trying to make call I get "Media stream permission denied" |
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| 11:13.18 | wasanzy | any help? |
| 11:13.34 | MacroMan | wasanzy, What SSL certificate are you using? |
| 11:13.39 | MacroMan | Have you enabled SSL? |
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| 11:16.00 | MacroMan | wasanzy, See https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial for information on making a secure connection |
| 11:30.18 | MacroMan | I'm getting lost audio in call recordings, but not in the calls themselves. What could cause that? |
| 11:35.23 | wasanzy | MacroMan: we have purchased an ssl cert. |
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| 11:44.04 | MacroMan | wasanzy, Please paste the output of 'http show status': sudo asterisk -x "http show status" | nc termbin.com 9999 |
| 11:44.58 | wasanzy | termbin.com? |
| 11:45.05 | MacroMan | terminal pastebin |
| 11:45.19 | wasanzy | http://termbin.com/0le0 |
| 11:45.38 | MacroMan | Your HTTPS server is not enabled |
| 11:46.37 | wasanzy | little confused. how do I enabled it then? because I have enabled it in sip and in my httpd config too |
| 11:47.24 | MacroMan | Here is mine for reference: https://paste.ngx.cc/4ea04690c5e8bfec |
| 11:47.27 | MacroMan | http.conf |
| 11:49.18 | wasanzy | ah I think you are right |
| 11:50.11 | wasanzy | this is all I have https://paste.linux.community/view/0a76b645 |
| 11:50.31 | MacroMan | You need to add the tls lines like in my conf |
| 11:50.36 | wasanzy | I was following this: https://kunjans.wordpress.com/2015/01/09/web-sip-client-sipml5-with-asterisk-13-on-centos-6-6/ |
| 11:51.56 | MacroMan | Funny, they go through the trouble of setting up tls on SIP, but don't show you how to enable it on http *sigh* |
| 11:52.23 | MacroMan | You're almost there, just configure tls in http.conf like I have and you'll be there. |
| 11:52.59 | wasanzy | ok |
| 11:53.03 | wasanzy | am doing that |
| 11:53.31 | wasanzy | I will have to generate tlsprivatekey? |
| 11:53.44 | MacroMan | You should have it from purchasing a cert |
| 11:54.02 | wasanzy | ok |
| 11:54.46 | wasanzy | well the one I purchased is what am using for my domains, is a wild card cert |
| 11:54.58 | MacroMan | Should work fine. |
| 11:55.05 | wasanzy | ok |
| 11:55.59 | MacroMan | Asterisk doesn't mind what you use as long as it's in a valid format. It's the client browser that cares |
| 11:57.05 | wasanzy | I have .crt .key and .crt(ca-bundle) |
| 11:57.20 | MacroMan | I think you need to convert to pem format |
| 11:57.28 | wasanzy | right ok |
| 11:57.39 | wasanzy | I will use asterisk command to convert it |
| 11:57.50 | MacroMan | good plan |
| 11:58.25 | MacroMan | The certificate file should be a chain also containing the ca-bundle |
| 11:58.40 | wasanzy | ok |
| 11:58.48 | wasanzy | I will get back when am done |
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| 12:34.14 | wasanzy | MacroMan: http://termbin.com/768z |
| 12:35.34 | wasanzy | should the keys match? I mean the certs on asterisk and the client(browser) |
| 12:38.45 | MacroMan | Yes. You have enabled the https server |
| 12:39.08 | MacroMan | Try browsing to your https address |
| 12:39.27 | wasanzy | MacroMan: I still get "Disconnected: Failed to connet to the server: |
| 12:39.38 | MacroMan | Try browsing to your https address |
| 12:39.46 | MacroMan | What do you get? |
| 12:40.05 | wasanzy | I have been able to browser but can't login |
| 12:40.23 | MacroMan | So you get an error free response when browsing? |
| 12:41.06 | wasanzy | I have been able to browser even without enabling the https sever. the main problem is the error "Disconnected: Failed to connet to the server:" |
| 12:41.08 | MacroMan | Well you should get a Not Found, but not a cert error |
| 12:41.21 | wasanzy | I get that when I hit login |
| 12:41.31 | MacroMan | Try browsing to your httpS address <------ Note the S |
| 12:41.46 | wasanzy | yes I am using https |
| 12:41.54 | MacroMan | You couldn't have browsed to it before, it wasn't enabled |
| 12:42.17 | wasanzy | MacroMan: no I was able to browser to it. |
| 12:42.28 | MacroMan | How? It wasn't enabled! |
| 12:42.35 | wasanzy | the problem am having is not browsing but login |
| 12:42.51 | MacroMan | "Failed to connet to the server" is not a login problem |
| 12:43.14 | wasanzy | ok |
| 12:43.36 | MacroMan | First try browsing to it. This will help you diagnose if there are anyu SSL/TLS problems |
| 12:43.52 | MacroMan | I assume you used port 7443 for tls? |
| 12:43.59 | wasanzy | from the sipML5 interface, I get that error when I hit login |
| 12:44.12 | MacroMan | OK. That's a connection problem |
| 12:44.37 | MacroMan | For example, you can browse to mine: https://sip.aplpackaging.co.uk:7443/ |
| 12:44.49 | wasanzy | ah sorry I think this is why: "wss://192.168.10.39:8088/wss |
| 12:45.42 | MacroMan | Yes. But do try these things by browsing to them first, you get much better error information |
| 12:45.53 | wasanzy | ok |
| 12:46.02 | wasanzy | let me try |
| 12:46.43 | wasanzy | I get "Not Found" |
| 12:47.01 | MacroMan | And you're on https and you can see you have a valid cert? |
| 12:47.34 | MacroMan | Then congratulations. All configured. Simply use the correct address in sipml5 |
| 12:47.56 | wasanzy | I tried this: https://IP:7443/ws and I get Upgrade Required |
| 12:47.56 | wasanzy | (null) |
| 12:47.56 | wasanzy | Asterisk Server |
| 12:48.07 | MacroMan | Take off the ws |
| 12:48.28 | wasanzy | MacroMan: I get "Not found" when I take off the ws |
| 12:48.30 | MacroMan | But it sounds like it's working |
| 12:49.18 | MacroMan | So use wss://IP:7443/ws in sipml5 and it should now work |
| 12:49.42 | wasanzy | that is what am using and it doesn't work |
| 12:50.07 | MacroMan | Have you set the WebSocket Server URL in sipml5 expert settings? |
| 12:50.13 | wasanzy | ah, connected |
| 12:51.05 | MacroMan | Brilliant. |
| 12:51.05 | wasanzy | C-00000000]: chan_sip.c:10387 process_sdp: Rejecting secure video stream without encryption details: video 3908 UDP/TLS/RTP/SAVPF 100 101 116 117 96 |
| 12:51.14 | wasanzy | whe I tried making call |
| 12:51.36 | MacroMan | I'm not sure, I've never used it for video befire |
| 12:51.44 | MacroMan | I only use audio |
| 12:51.47 | wasanzy | ah sorry, I have just disabled video, let me try |
| 12:54.46 | MacroMan | You've got fun times ahead. I've just finished a month long project developing a softphone in sipjs |
| 12:55.34 | wasanzy | that sound interesting |
| 13:05.40 | *** join/#asterisk Haris (~haris@unaffiliated/haris) |
| 13:09.51 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
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| 13:31.29 | *** part/#asterisk Haris (~haris@unaffiliated/haris) |
| 13:39.02 | wasanzy | I am able to make call from my sipML5 but I can't hear any sound. my cli logs : https://paste.linux.community/view/14e0103a |
| 13:42.08 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 13:43.06 | Samot | Where's an actual, full call? |
| 13:43.45 | wasanzy | Samot: that is a full call unless you are talking about SIP debug |
| 13:44.00 | Samot | Yes |
| 13:44.17 | wasanzy | ok, am getting thatr |
| 13:46.50 | wasanzy | https://paste.linux.community/view/c3f13e7b |
| 13:47.03 | stefan27 | If you're gonna work with sipml for a while, you might want to start a wireshark session on your pc, as well as log into chrome://webrtc-internals |
| 13:47.15 | stefan27 | to see if rtp and sip packets are flowing correctly |
| 13:48.08 | stefan27 | the chrome://webrtc-internals can be used to gather data about the peerconnection, which includes info about the audio input from your microphone |
| 13:48.40 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-zxhekuhtggaaguvr) |
| 13:49.47 | Samot | Contact: "6000"<sips:6000@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr" |
| 13:50.06 | Samot | set_destination: Parsing <sips:6000@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss> for address/port to send to |
| 13:50.06 | Samot | set_destination: URI is for WebSocket, we can't set destination |
| 13:50.11 | Samot | Fix your softphone. |
| 13:51.09 | wasanzy | you mean I should disable RTCWeb Breaker? |
| 13:51.13 | Samot | Your contact contains a bad URI |
| 13:51.49 | Samot | sips:6000@df7jal23ls0d.invalid <--- Not valid. |
| 13:52.16 | wasanzy | Samot: that is not from my phone, I don't know where it is from |
| 13:52.50 | Samot | Who's 6000? You? |
| 13:52.55 | Samot | Who's making this call? |
| 13:53.07 | MacroMan | Funny, when I tested with sipml5, I had to force rtcweb breaker on |
| 13:53.29 | MacroMan | In the expert settings of the demo app |
| 13:54.01 | wasanzy | MacroMan: I have it on as well |
| 13:54.13 | Samot | Ext 6000 is making the call, correct? |
| 13:54.28 | stefan27 | rtcweb breaker attribute does nothing unless your sipml client talks to an intermediate gateway from doubango |
| 13:54.44 | stefan27 | if your sipml client registers directly with asterisk, asterisk will just ignore that value |
| 13:54.48 | wasanzy | Samot: yes but even when I don't make call, I see those funny contacts in the log |
| 13:54.58 | Samot | OK |
| 13:55.06 | Samot | In regards to THIS call, that is the issue. |
| 13:55.20 | Samot | Where else do you see those? Are people using this softphone? |
| 13:55.50 | wasanzy | no one else is using it |
| 14:05.38 | *** join/#asterisk SupaYoshi (~SupaYoshi@104.223.1.186) |
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| 14:13.41 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 14:53.01 | wasanzy | The currently supported voice codecs are G.711, G.722, iLBC |
| 14:53.11 | wasanzy | I think that is my problem |
| 15:04.41 | *** join/#asterisk ruied (~ruied@81.84.234.209) |
| 15:11.12 | *** join/#asterisk kharwell (kharwell@nat/digium/x-nzobpkeevzfmuzyv) |
| 15:11.12 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:11.13 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 15:15.39 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 15:22.09 | ruied | How can I send a text message to a phone without dial to it? is it possible. When I receive an outside call it reaches sip/300 but I would like to know if I can send just a CALLERID(num) to sip/301 so sip/301 could pickup the call if he wants, via pickupgroup. |
| 15:23.27 | igcewieling | you might be able to hack something up to accomplish that, but it will be so complicated and time consuming it will not be worth it. |
| 15:24.01 | igcewieling | 1) you CANNOT do dialplan stuff without a call. 2) you can generate calls using methods other than telephones. |
| 15:24.30 | igcewieling | ruied: have you look at "core show applications" and "core show functions" ? You should. |
| 15:25.31 | ruied | ok. |
| 15:30.22 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 15:31.26 | igcewieling | You will find an application called 'MessageSend' |
| 15:33.30 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 15:38.32 | MacroMan | Recommendations for free hold music? |
| 15:39.03 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 15:39.41 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |
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| 15:43.33 | *** join/#asterisk chendy (~alexc@121.34.144.88) |
| 15:45.38 | igcewieling | MacroMan: I recommend the free hold music included with Asterisk. |
| 15:45.59 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-affebwtjhahywfbl) |
| 15:45.59 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 15:46.08 | igcewieling | There are also a bazillion royalty free music on hold sites. |
| 15:47.21 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:50.35 | MacroMan | Of course. I was hoping someone may have pointed me in the direction of something good rather than having to trawl dozens of sites trying to find what others have already found. |
| 15:53.00 | ruied | MacroMan, I have used this one yesterday: https://www.jamendo.com/. You might need license... |
| 15:59.42 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:01.54 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 16:05.56 | wasanzy | still not getting any sound |
| 16:07.16 | wasanzy | https://paste.linux.community/view/8226243b |
| 16:07.48 | MacroMan | After changing moh files, what command do I need to issue to reload? 'module reload res_musiconhold.so' doesn't work, I get errors about missing files |
| 16:08.03 | MacroMan | I don't want to hard restart |
| 16:12.20 | Samot | set_destination: Parsing <sips:6000@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss> for address/port to send to |
| 16:12.20 | Samot | set_destination: URI is for WebSocket, we can't set destination |
| 16:12.26 | ruied | MacroMan, moh <tab> |
| 16:12.48 | Samot | ^^^ Same error, not sure how fixing codec support helps the URI being parsed. |
| 16:13.14 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 16:13.30 | Samot | wasanzy: You're not getting any sound because the call is ending. |
| 16:13.39 | MacroMan | ruied, That's just shorthand for 'module reload res_musiconhold.so', which doesn't work |
| 16:13.51 | MacroMan | (I've just tried) |
| 16:13.59 | wasanzy | Samot: any suggestion how to set the destination? I really don't know where that is coming from |
| 16:14.12 | Samot | Contact: "6000"<sips:6000@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr" |
| 16:14.18 | Samot | ^^ Same as before. |
| 16:14.45 | wasanzy | I get this in the log: <SIP/6000-0000000f> Playing 'enquiries.g722' (language 'en') |
| 16:15.00 | Samot | Try not playing a g722 file. |
| 16:15.27 | wasanzy | Samot: I tried ulaw and alaw |
| 16:15.32 | wasanzy | same thing |
| 16:15.44 | Samot | Is this only happening on WSS calls? |
| 16:16.09 | wasanzy | Samot: yes |
| 16:16.27 | Samot | Have you tried a different webrtc client? |
| 16:16.49 | wasanzy | what other client can I try? zoiper? |
| 16:17.09 | MacroMan | Good point. I personally did have luck wioth sipml5, but you could try sipjs |
| 16:17.19 | MacroMan | Which is what I use in production now |
| 16:17.30 | MacroMan | https://sipjs.com/demo-phone/ |
| 16:17.30 | *** join/#asterisk freebs (~freebs@unaffiliated/freebs) |
| 16:19.02 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 16:19.19 | Samot | Zoiper is a softphone not a webrtc client. |
| 16:19.33 | Samot | But if you try zoiper you have to use a non-ws/wss protocol. |
| 16:20.04 | wasanzy | MacroMan: ok I will try that |
| 16:20.21 | wasanzy | Samot: yea, just to make sure the sound works |
| 16:20.31 | wasanzy | could it be a firewall problem? |
| 16:21.03 | wasanzy | I realized the sound doesnt work when am on this current network using other sipphones too |
| 16:21.07 | Samot | You can try and see. |
| 16:21.14 | MacroMan | Disabled it and see |
| 16:21.25 | MacroMan | What Samot said |
| 16:22.20 | wasanzy | ok |
| 16:25.32 | igcewieling | If it wasn't for customers life would be good. Our Tech: "I hope all is well. I have checked your system and I see that Extension 10's voicemails are being emailed to nyorders@blah.com." Client: "Which one is my modem?" |
| 16:25.51 | MacroMan | This makes no sense. I've completely restarted * and it's still trying to find old moh files: https://paste.ngx.cc/d98f3c90dc6ccf31 |
| 16:26.58 | igcewieling | Are you specifying the MoH files or the MoH directories in your moh config? |
| 16:27.23 | MacroMan | That will be it |
| 16:28.11 | igcewieling | that doesn't answer my question. |
| 16:28.41 | MacroMan | Well files, but changing to directories gives the error res_musiconhold.c:1329 _moh_register: Don't know how to do a mode 'directories' music on hold |
| 16:29.01 | igcewieling | you cannot specify moh files, only directories |
| 16:29.09 | igcewieling | You have some other issue |
| 16:29.28 | igcewieling | pastebin your musiconhold.conf and hope someone wants to help |
| 16:29.36 | *** join/#asterisk sarthor (~sarthor@unaffiliated/sarthor) |
| 16:29.38 | wasanzy | well using zoiper with a different sip account, the sound work. so I guess the problem is with the wss |
| 16:29.44 | sarthor | Hi, I have Dlink ip phone DPH400SE, I want connect this IP phone with my Openvpn server, while it support Openvpn, but I am unable to find any manual, how to upload openvpn config files. I clicked on brows, and tried to upload, but says error, please let me know if some one knows, how to configure IP Phone for Openvpn. Thanks. |
| 16:30.00 | MacroMan | musiconhold.conf: https://paste.ngx.cc/496d45b615a85be3 |
| 16:30.06 | MacroMan | Very, very simple |
| 16:30.19 | igcewieling | sarthor: I doubt anyone here will help you setup VPNs |
| 16:31.20 | igcewieling | are the files in /var/lib/asterisk/moh and are they readable by the asterisk user. |
| 16:31.26 | MacroMan | yes |
| 16:31.29 | ruied | sarthor, #openvpn channel |
| 16:32.02 | igcewieling | then you have some other problem. what format are the files in? |
| 16:32.12 | MacroMan | I added a new file and removed the ones I didn't want. Those errors in https://paste.ngx.cc/d98f3c90dc6ccf31 are the files I've removed |
| 16:32.13 | sarthor | igcewieling: ruied thanks. |
| 16:32.16 | MacroMan | wav |
| 16:32.33 | MacroMan | Files in the directory played fine before I changed them for another file |
| 16:33.03 | Samot | ls -l /var/lib/asterisk/moh |
| 16:33.15 | igcewieling | that means nothing. use the "file" command. like this: moh]# file macroform-cold_day.wav |
| 16:33.15 | igcewieling | macroform-cold_day.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 16:33.42 | MacroMan | http://termbin.com/6wtr |
| 16:34.05 | igcewieling | if it says something like 16bit stereo 44000khz then your |
| 16:34.15 | igcewieling | files are in the wrong format |
| 16:34.20 | MacroMan | Asterisk has rememered what was in the directory and is still trying to play that, even after a restart |
| 16:34.26 | Samot | Uhm... |
| 16:34.34 | Samot | -rw-r--r-- 1 root root 4464220 Dec 26 2009 macroform-the_simplicity.wav.no |
| 16:34.37 | igcewieling | MacroMan: it is not. |
| 16:34.44 | Samot | ^^^ might be why the .wav file can't be found. |
| 16:34.55 | MacroMan | It is: https://paste.ngx.cc/d98f3c90dc6ccf31 |
| 16:35.06 | MacroMan | I want to play carefree.wav |
| 16:35.19 | MacroMan | Hence why I've renamed the others .wav.no |
| 16:35.22 | Samot | [Mar 10 16:24:34] WARNING[1728][C-00000001]: res_musiconhold.c:365 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh/carefree': No such file or directory |
| 16:35.36 | igcewieling | well that was easy. your file is in the WRONG FORMAT. |
| 16:35.56 | igcewieling | "]: format_wav.c:105 check_header_fmt: Not in mono 2" |
| 16:36.17 | igcewieling | your files MUST be in 16 bit, mono 8000 Hz or asterisk will not be able to read it. |
| 16:36.24 | Samot | Because he change those files names |
| 16:36.37 | igcewieling | Samot: moh won't play thos files. |
| 16:36.40 | Samot | Those are going to error out because the files don't exist... |
| 16:36.42 | Samot | Right. |
| 16:36.47 | Samot | That's not what he's concerned about. |
| 16:36.49 | igcewieling | it will only play the single non-no file he has.. |
| 16:36.55 | Samot | I know |
| 16:36.56 | igcewieling | the carefree one. |
| 16:36.57 | Samot | He knows. |
| 16:36.58 | MacroMan | OK I see. But why is still trying to play the old files? |
| 16:37.03 | Samot | Which it's not playing. |
| 16:37.08 | Samot | the carefree is the one he wants to play. |
| 16:37.12 | *** part/#asterisk tuxian (~tuxian@igilmour.plus.com) |
| 16:37.13 | Samot | [Mar 10 16:24:34] WARNING[1728][C-00000001]: res_musiconhold.c:365 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh/carefree': No such file or directory |
| 16:37.22 | igcewieling | MacroMan: try removing them. |
| 16:37.42 | igcewieling | that is a misleading message |
| 16:37.54 | Samot | Show the moh conf |
| 16:38.55 | MacroMan | <MacroMan> musiconhold.conf: https://paste.ngx.cc/496d45b615a85be3 |
| 16:39.08 | ruied | ps aux |grep asterisk is there more than one * running?? |
| 16:39.26 | MacroMan | igcewieling, Thanks, removing those fixed the other errors. I'll change the format of the fiule |
| 16:40.45 | Samot | Now show the dailplan that calls the moh |
| 16:40.55 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 16:41.53 | MacroMan | Samot, No. There is nothing wrong with the dialplan, I simply have the wav file in the wrong format |
| 16:42.07 | Samot | OK |
| 16:43.06 | MacroMan | igcewieling, Are you sure I have to have the sample rate as low as 8000? I though * could play 16000? |
| 16:43.51 | igcewieling | If you want to use WAV, then yes. |
| 16:44.27 | igcewieling | if you can use something g722 or sln16 as your moh fileformat, you should be able to get a higher sampling rate |
| 16:45.02 | igcewieling | or you could just live with 8000 since that's all any non-IP phone can handle. |
| 16:45.25 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
| 16:45.38 | MacroMan | OK thanks |
| 16:46.53 | WIMPy | G.722 exists on cordless phones as well. |
| 16:47.49 | WIMPy | A few month ago I had my first G.722 call to another carrer BTW. \o/ |
| 16:48.49 | *** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 16:49.05 | MacroMan | Hmm I'm struggling with this. I used Audicty to change the but depth and sample rate, but * is still not happy |
| 16:49.11 | MacroMan | <PROTECTED> |
| 16:49.34 | WIMPy | Make sure the file doesn't contain any tags. |
| 16:49.52 | igcewieling | MacroMan: try renaming the file to whatever.ulaw and see if that helps. WAV files need headers, ulaw files shouldn't |
| 16:50.13 | igcewieling | WIMPy: which carrier? sprint? |
| 16:50.25 | WIMPy | That's a different sample format, though. |
| 16:50.35 | WIMPy | From O2 to Vodafone. |
| 16:50.49 | igcewieling | Ah, you are in the civilized part of the world. |
| 16:51.27 | WIMPy | The part that feeds from the civilized times. |
| 16:51.40 | igcewieling | Many aspects of telecom in the USA are very backward. |
| 16:51.53 | MacroMan | igcewieling, Ouch my ear. I renamed it .ulaw and I got very loud grey noise |
| 16:52.13 | WIMPy | Yes, that's the main thing I learned since using Asterisk. |
| 16:52.21 | igcewieling | MacroMan: drat. many people use sox to do the conversions. |
| 16:52.42 | MacroMan | I'll see what options I have in audicity. I may be able to export with a header |
| 16:52.46 | WIMPy | uses audacity. |
| 16:53.22 | WIMPy | Just export as wav and make sure all fields for tags are empty. |
| 16:54.08 | MacroMan | Strange, I exported with the same options and it works this time. |
| 16:54.18 | MacroMan | Thanks for your help |
| 16:54.32 | sarthor | What is most easy solutions if ISP/Country blocking voip. |
| 16:55.03 | WIMPy | Define "blocking voip". |
| 16:55.37 | WIMPy | heard it's often just a filter for udp/5060. So that's easy. |
| 16:55.49 | WIMPy | With many options. |
| 16:57.26 | sarthor | WIMPy: I can connect extentions with in local network, but I am unable to do so via Internet. |
| 16:57.51 | WIMPy | That can have many reasons. |
| 16:59.35 | MacroMan | Odd one. Can I configure hold so I can still hear the other party? Like mute, only they hear moh |
| 16:59.45 | Samot | No. |
| 16:59.54 | Samot | Hold changes the RTP flow to one way |
| 17:00.24 | MacroMan | Shame. I have noticed that their channel is recorded in Monitor when on hold though |
| 17:00.26 | Samot | Changes the RTP to sendonly. |
| 17:00.49 | Samot | Because that channel has MoH on it. |
| 17:01.03 | sarthor | WIMPy: nmap -p 188.51.174.32 says 5060/tcp filtered sip, Is it Opened? |
| 17:01.13 | MacroMan | Hmm. I could run a macro to mute our end and play with Playback |
| 17:01.34 | WIMPy | sarthor: Sip is usually UDP. |
| 17:02.18 | sarthor | how to do nmap for udp ;) I am newbie |
| 17:02.32 | MacroMan | Good weekend peeps |
| 17:03.41 | WIMPy | It's not thtat easy to scan for UDP ports. You can only hope the destination dends ICMP messages when there's noone listening. |
| 17:04.17 | WIMPy | Just sniff what happens when you try to register a phone. |
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| 17:50.50 | wasanzy | any idea how I can fix this? set_destination: URI is for WebSocket, we can't set destination |
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| 18:32.07 | adnauseum | sarthor: man udp; -sU |
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| 19:21.48 | canthus13 | I have calls being dropped shortly after connecting. According to the company we're trunking through, they're seeing 403 and 500 errors on the ladder diagram. Some calls show up in the full log and appear like the server is simply hanging up. others I'm not seeing in the log at all. Any ideas of what might be occurring? The host company upgraded the firmware on an edge router last night in an attempt to mitigate this and voice quality i |
| 19:23.17 | Chainsaw | canthus13: Is RTP establishing both ways? |
| 19:23.38 | Chainsaw | canthus13: Don't look at SIP alone, you need more then that for a successful call. |
| 19:23.58 | canthus13 | Not sure. They didn't include the diagrams and I can't get ahold of the tech on the ticket at the moment. He went to lunch. :/ |
| 19:24.05 | Samot | Well 403 is a forbidden error. |
| 19:24.13 | Samot | Which means something wasnt allowed to do what it wanted. |
| 19:24.17 | canthus13 | Right. It's giving forbidden/security check failed. |
| 19:24.21 | Samot | 500 is a general server side error. |
| 19:24.41 | Samot | Need to see an actual SIP debug from Asterisk of the call(s) |
| 19:25.20 | canthus13 | Hmm... I've got tcpdump running in hopes of catching it in action, but it's been intermittent. |
| 19:27.08 | igcewieling | canthus13: does the call disconnect before ANY audio is happening? |
| 19:27.32 | canthus13 | Nope. They'll get a couple of seconds into the call and it just goes dead. |
| 19:27.38 | igcewieling | disable directmedia |
| 19:27.47 | igcewieling | that is the classic symptom |
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| 19:28.39 | canthus13 | I'll test that on a couple of servers and see how it goes. |
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| 19:38.14 | canthus13 | Disabling directmedia just involves directmedia=no added to the peer details? |
| 19:39.16 | igcewieling | I'd put it in [general] unless most of your peers are not behind NAT |
| 19:39.26 | canthus13 | ok |
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| 21:04.34 | canthus13 | Ok. So. I finally got info from the ESBC for a call. I'm not quite sure what I'm looking at, but it seems like RTP is the problem. |
| 21:04.38 | canthus13 | http://pastebin.com/DaQzBF66 |
| 21:14.41 | igcewieling | the SDP has your private IP address in it. It shouldn't. |
| 21:15.28 | igcewieling | look at localnet= and externip= and externhost= settings in chan_sip.conf, if you are using chan_pjsip then I can't help. |
| 21:16.00 | canthus13 | no pjsip. (Although that broke a server once... So i'm aware of it, at least. :) |
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| 21:17.32 | igcewieling | if you send an invite with an SDP pointing audio at an IP like 172.16.0.5, your provider is going to have a hard time sending audio to that IP. Assuming you don't have a private interconnect with your carrier. |
| 21:18.41 | canthus13 | We do. It's hosted with the carrier. I've not seen that particular IP, though. as far as I was told, the ESCB is at a different IP. |
| 21:19.35 | igcewieling | When rfc1918 addresses are involved, things get tricky very quickly. |
| 21:21.24 | canthus13 | Thing is this has worked for several years without problems. But to add to the complexity, there are VPN tunnels connecting all the sites that internal calls are routed through. I'm wondering if there isn't address duplication somewhere that's causing issues... |
| 21:22.03 | igcewieling | Does anyone else schedule downtime but not do anything just to see which customers complain and claim the maintenance broke something even though nothing was done? |
| 21:23.13 | canthus13 | Ha. I like that idea. Our net admin plugged a switch into itself today just to see people freak out... |
| 21:24.45 | igcewieling | I don't do it often, but it can be a very instructive exercize. |
| 21:25.11 | igcewieling | er.. exercise |
| 21:27.51 | canthus13 | classic BOFH... |
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