| 01:00.49 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:34.01 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 01:55.48 | *** join/#asterisk noko (pavel@gate6.zhovner.com) |
| 01:56.58 | Gerbilest | hello everyone, can I get some help with the simple 2 phone example pbx if anyone has a moment? |
| 01:57.24 | noko | I use Queue() in my dial plan. How to prevent answering until someone in queue really answer? |
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| 01:59.13 | [TK]D-Fender | noko, "core show application queue" <- |
| 01:59.41 | [TK]D-Fender | Gerbilest, Show us where your calls have gotten, and what' you've done and we'll see what we can advise. |
| 01:59.48 | [TK]D-Fender | Gerbilest, PASTEBIN is your friend. |
| 01:59.51 | [TK]D-Fender | ~pb |
| 01:59.51 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 01:59.53 | [TK]D-Fender | ^^^ |
| 02:00.00 | Gerbilest | I know what pastebin is thanks, |
| 02:00.10 | Gerbilest | I'm just not showing the two users in sip show peers |
| 02:00.15 | Gerbilest | though it appears that they've registered |
| 02:00.20 | Gerbilest | on the softphone |
| 02:01.16 | noko | [TK]D-Fender, Ring instead of playing MOH? |
| 02:01.22 | Gerbilest | it shows zero sip peers |
| 02:01.33 | [TK]D-Fender | Gerbilest, Show us the status dumps, and what you've configured |
| 02:01.47 | [TK]D-Fender | noko, Nope, keep reading.... |
| 02:02.01 | Gerbilest | https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts |
| 02:02.05 | Gerbilest | it's configured exactly like that |
| 02:03.01 | Gerbilest | the service was restarted and then I've attempted to register the softphone (zoiper) with the demo-bob account |
| 02:03.01 | noko | [TK]D-Fender, I can't find :( |
| 02:03.30 | Gerbilest | the softphone will show the account as registered, though I have 0 sip peers in the show sip peers readout |
| 02:03.35 | [TK]D-Fender | noko, I'm missing something myself... I recall there being an option for this... |
| 02:03.47 | JeffC_NN | I think noko wants the incoming call to not be answered, aka Answer(), until there's an agent ready to talk |
| 02:03.51 | [TK]D-Fender | Gerbilest, Show us your actual configs and debug |
| 02:04.22 | hfp | Hey all, I'm looking for this website with some fake guy called Bernie or something like that who would drive telemarketers crazy... It was all done with an Asterisk dialplan and you could listen to the latest calls because it would record them automatically |
| 02:04.24 | hfp | Does that ring a bell for anyone? Can't find the website anymore |
| 02:04.58 | [TK]D-Fender | noko, "R" I think actually is it... |
| 02:05.12 | [TK]D-Fender | noko, nothing but "riung", and I don't believe that implies in-band, so that should do it. |
| 02:05.51 | [TK]D-Fender | noko, Go try it. If it's a SIP call it's easily provable when the answer actually happens. Show us a call with "sip set debug on" if applicable and you're not sure |
| 02:06.00 | [TK]D-Fender | Gerbilest, Show us |
| 02:07.15 | Gerbilest | There are no debug messages and I literally copied and pasted the configs in the web address. |
| 02:07.25 | Gerbilest | that I linked to earlier. |
| 02:07.51 | Gerbilest | at least no debug messages on -vvvvv |
| 02:07.53 | noko | [TK]D-Fender, it play ring sound but answer immediately |
| 02:08.05 | [TK]D-Fender | I want to see actual configs,actual CLI commands called. Complete real dump. Including a reload of the module itself\ |
| 02:08.57 | [TK]D-Fender | noko, Show us |
| 02:10.33 | noko | [TK]D-Fender, I tested it by calling from my phone and see that answer timer is start |
| 02:10.42 | [TK]D-Fender | noko, Show us the call. |
| 02:11.19 | noko | aww~ debug log is so big. Okay -_- |
| 02:11.28 | [TK]D-Fender | Logs are worthless. |
| 02:11.29 | JeffC_NN | tail ;) |
| 02:11.30 | [TK]D-Fender | Live CLI only |
| 02:12.00 | [TK]D-Fender | "asterisk -r", "core set verbose 10", "sip set debug on" |
| 02:12.25 | noko | how to filter debug only by one peer? |
| 02:12.36 | JeffC_NN | can't :/ |
| 02:12.50 | noko | because it's very high traffic here |
| 02:13.04 | Gerbilest | WHOIS [TK]D-Fender |
| 02:13.06 | noko | few screens of logs in one second |
| 02:13.27 | Gerbilest | whoops lol |
| 02:13.34 | [TK]D-Fender | Gerbilest, I'm me. You're you. That's how this works. |
| 02:13.44 | Gerbilest | yeah that's great |
| 02:13.48 | Gerbilest | thanks for the help |
| 02:14.16 | JeffC_NN | noko, mabye it would help if you had a test install of asterisk? |
| 02:14.21 | [TK]D-Fender | noko, I have no issues scrolling through. |
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| 02:21.09 | noko | [TK]D-Fender, http://pastebin.com/raw.php?i=y2ky5h7h |
| 02:22.47 | [TK]D-Fender | noko, your "tT" options BREAK this |
| 02:23.12 | [TK]D-Fender | noko, Asrterisk can't listen for DTMF from the caller if it isn't listening. Listening means answering |
| 02:23.17 | [TK]D-Fender | remove those |
| 02:23.20 | [TK]D-Fender | retest |
| 02:25.30 | noko | [TK]D-Fender, same result http://pastebin.com/raw.php?i=qAqxRpJA |
| 02:28.05 | [TK]D-Fender | I see another issue there... this option mentions ringing when calling the agent... is says nothing about the period BEFORE calling... |
| 02:28.09 | [TK]D-Fender | And we see MoH starting. |
| 02:28.32 | [TK]D-Fender | try removing the class from your queue definition and make sure there is no [general] on assigned either |
| 02:29.08 | [TK]D-Fender | I thing "r" might compliment "R" as well... |
| 02:29.20 | [TK]D-Fender | Do both, make those other changes and retest |
| 02:30.06 | noko | [TK]D-Fender, here is my queues.conf http://take.ms/Jhlzq |
| 02:30.41 | [TK]D-Fender | looks fine |
| 02:30.49 | [TK]D-Fender | ok, test with "rR" |
| 02:31.10 | noko | same result |
| 02:31.27 | [TK]D-Fender | :/ |
| 02:31.51 | [TK]D-Fender | Was pretty sure there's a way for this... but looking at my own system I don't see anything more definitive. |
| 02:33.06 | JeffC_NN | How would * bridge the agent to the call that isn't answered unless app_queue answers it? If all else fails, a custom AGI would work. |
| 02:33.37 | noko | [TK]D-Fender, lol, i'm sorry. My dialplan starts from Answer() and only than Queue() |
| 02:33.43 | noko | working now |
| 02:33.53 | noko | [TK]D-Fender, thank very much! |
| 02:34.10 | JeffC_NN | Yay! |
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| 02:39.19 | [TK]D-Fender | .... |
| 02:39.36 | [TK]D-Fender | noko, Next time, show it from the START |
| 02:39.44 | Gerbilest | wow. |
| 02:39.46 | [TK]D-Fender | noko, That was a pretty huge mistake |
| 02:39.50 | [TK]D-Fender | :) |
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| 08:27.38 | R4v3n | hello :) Somebody does H.323 video calls through Asterisk please ? Thank you ! |
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| 10:55.49 | vjosh | hi |
| 10:57.24 | vjosh | whois bsdice |
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| 10:58.27 | vjosh | need help |
| 10:58.35 | vjosh | in asterisk ARI |
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| 13:53.04 | wonderworld | whats your fav tool to debug sip traffic on console? is there a specialised tool that makes the job a bit more easy? i really like wiresharks sip plugin and are looking for something similar in text mode. |
| 13:54.24 | WIMPy | sngrep |
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| 15:24.13 | angryuser | Hi, after reading the article of the https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway i still do not understand if the gateway is able to originate faxes ? can i send tiff files like with the digium T.38 module ? |
| 15:24.56 | angryuser | something like exten => 1,n,Sendfax(myreceivedfile.tiff,d) |
| 15:28.45 | malcolmd | res_fax_digium doesn't do gateway. gateway is when you want to go from PSTN directly to T.38 on a SIP call without store-and-forward. if you're calling SendFax, then gateway isn't being invoked. if you need gateway, you need res_fax_spandsp |
| 15:30.10 | newtonr | angryuser, yes you can send faxes using Sendfax |
| 15:31.04 | angryuser | so still the oldy spandsp for the termination |
| 15:32.20 | angryuser | so today the best option is still the digium t.38 module for sending faxes |
| 15:32.38 | angryuser | (origination from asterisk and tiff file) |
| 15:33.10 | malcolmd | either will work, res_fax_digium or res_fax_spandsp, for that purpose |
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| 15:36.34 | newtonr | angryuser, I recommend the destruction of all fax machines in your proximity |
| 15:36.51 | malcolmd | the realtors won't stand for that |
| 15:37.04 | newtonr | Out with the realtors! |
| 15:37.06 | angryuser | newtonr: the problem are the people not poor fax machines |
| 15:37.43 | angryuser | i wonder if the digiums one will work for the 13.1 as it is listed to 12 in the downloads |
| 15:39.37 | *** join/#asterisk RadJackson (~RadJackso@cpe-xd001622.cust.jaguar-network.net) |
| 15:43.35 | RadJackson | Hello , i'm having two asterisk server , from my cellphone i call Server A that has to transfer the call to B. on server A i've created the SIP account (trunk) , Server A configuration is http://pastebin.com/WwbmqZg0 |
| 15:43.35 | RadJackson | Server B configuration is : http://pastebin.com/BHAh3Vgb |
| 15:43.35 | RadJackson | Everything works fine , expect i keep getting a notice on both server , res_hep.c:418 hep_queue_cb: Unable to send packet: Address Family mismatch between source/destination , any idea please ? |
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| 15:46.06 | newtonr | RadJackson, the message only happens during the successful transfer? |
| 15:46.23 | RadJackson | yes on both servers i see the same message only when the call is bridged |
| 15:47.05 | newtonr | Hmm, and it all works fine. |
| 15:48.46 | newtonr | I, have no idea. |
| 15:49.22 | RadJackson | yes |
| 15:49.42 | RadJackson | it all works fine i just keep getting that notice |
| 15:49.54 | newtonr | What versions of Asterisk? |
| 15:49.59 | RadJackson | [2014-12-23 16:49:09] NOTICE[19850]: res_hep.c:418 hep_queue_cb: Unable to send packet: Address Family mismatch between source/destination |
| 15:50.06 | RadJackson | Asterisk 13.1.0, |
| 15:50.10 | RadJackson | on both servers |
| 15:51.51 | RadJackson | can it be related to the type = user/peer/friend parameter? Server A can receive calls from outside and transfer them to server B , but server B will only send calls to server A , this is the actual architecture |
| 15:52.50 | newtonr | I'm really not sure. I've personally never seen that message before. Not that I can remember at least. |
| 15:53.52 | newtonr | I'd expect it to be about receiving IPv6 when expecting IPv4, or vice versa. |
| 15:54.20 | RadJackson | Thats exactly the only answer i see on forums "You probably received an IPv6 address when your system only supports IPv4." |
| 15:54.24 | RadJackson | Can i do something about this? |
| 15:54.34 | newtonr | A developer would need to look at the code around that notice and see why Asterisk prints it out. |
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| 15:55.35 | newtonr | RadJackson, if everything is working fine, you probably don't need to do anything. The fact that the NOTICE gets printed out may be the only problem. |
| 15:56.31 | newtonr | RadJackson, if you like, you can file a bug and include a packet capture of the call that results in the message, as well as an Asterisk full log including SIP trace and "DEBUG" logger messages. |
| 15:56.54 | RadJackson | ok |
| 15:57.42 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |
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| 16:40.58 | mjordan | RadJackson: are you using HEP? |
| 16:42.04 | mjordan | The HEP protocol, which res_hep is using, only supports encapsulating packets from the same address family. If the source address is IPv4 and the destination is IPv6 (or vice versa), it can't send the packet to the Homer server |
| 16:42.40 | mjordan | Of course, if you don't have a Homer server, you should just disable res_hep |
| 16:42.42 | mjordan | or unload it |
| 16:45.13 | RadJackson | mjordan thanks a lot |
| 16:45.19 | RadJackson | i've disabled ipV6 on my debian server |
| 16:45.23 | RadJackson | not having the notice anymore |
| 16:46.27 | newtonr | Did you have a mix of address families? Or did you just have IPv4 on both sides? |
| 16:47.20 | RadJackson | i had ipV6 activated on server A , desactivated it , not having the problem |
| 16:48.13 | RadJackson | just by updating a file located at /proc/sys/net/ipv6/conf/NETWORK INTERFACe , file name is /proc/sys/net/ipv6/conf , modified its content from 0 to 1 |
| 16:51.15 | mjordan | RadJackson: I am curious, are you using Homer? |
| 16:52.21 | RadJackson | nop what is that? |
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| 17:01.00 | mjordan | RadJackson: what res_hep is attempting to communicate with :-) |
| 17:01.29 | mjordan | RadJackson: rather than removing IPv6 from a server, I'd simply disable res_hep in hep.conf or unload res_hep. If you aren't using, you don't need the overhead that module incurs |
| 17:01.44 | mjordan | Homer: http://www.sipcapture.org/ |
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| 17:39.47 | RadJackson | mjordan ok i will try disabling res_hep |
| 17:39.50 | RadJackson | thanks a lot |
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| 17:40.36 | RadJackson | whats the purpose ot res_hep ? |
| 17:42.07 | newtonr | mjordan pretty much just explained that to you but: https://wiki.asterisk.org/wiki/display/AST/New+in+13#Newin13-res_hep |
| 17:42.16 | newtonr | "A new module, res_hep, has been added that acts as a generic packet capture agent for the Homer Encapsulation Protocol (HEP) version 3. It can be configured via hep.conf. Other modules use res_hep to send message traffic to a HEP capture server." |
| 17:43.23 | newtonr | I also recommend the link he gave you: http://www.sipcapture.org/ |
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| 17:54.04 | RadJackson | thanks |
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| 18:22.24 | newtonr | No problem! |
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| 19:29.43 | c0ldg0ld | What are people out here using to monitor their asterisk servers? zenoss, nagios, zabbix, something I haven't heard of yet? |
| 19:30.44 | c0ldg0ld | I have about 15 servers of different asterisk vintages (for now) that I've decided would be in my best interest to have monitored by some service to try and keep me ahead of irritated users. I figured here was the place to find up-to-date advice |
| 19:32.13 | c0ldg0ld | none internet facing and all on the same network though spread all over town, if that makes a difference |
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| 19:35.59 | [TK]D-Fender | Nagios is common. |
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| 19:36.09 | [TK]D-Fender | I've heard reference to Zabbix IIRC |
| 19:37.40 | c0ldg0ld | is Nagios probably the most used? I'm not terribly interested in a bunch of bells and whistles, just something that will let me know if a PRI has gone down or IAX trunk etc |
| 19:38.09 | [TK]D-Fender | Those are the kinds of thins people commonly check for |
| 19:38.28 | [TK]D-Fender | I'm sure you can Google up dozens of config samples for it in a matter of seconds |
| 19:39.37 | c0ldg0ld | Yeah, I started googling for zenoss because that's what a buddy said he used but just for linux admin, not Asterisk. The things I were noticing were for Ast 1.6 and were quite old so I didn't know if there was something better. |
| 19:39.40 | c0ldg0ld | Thanks for the input |
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| 20:01.18 | Kobaz | so i have a polycom that's continually asking for dhcp, and the server is giving sending back an address, but the phone's not taking it |
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| 20:07.29 | malcolmd | that's weird |
| 20:09.42 | zaf | is something else using it? |
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| 20:16.41 | Kobaz | never mind, fixed it |
| 20:16.43 | Kobaz | rogue dhcp server |
| 20:17.09 | MaliutaLap | gotta love that |
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| 22:50.43 | warewolf | anybody here good w/ SIP-TLS and SRTP? |
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| 22:51.37 | warewolf | I'm trying to get my asterisk box to Dial() out w/ them enabled, and it doesn't appear to work w/ the channel vars (secure_bridge_signaling and secure_bridge_media) unless I'm Dial()'ing a peer that is already defined. |
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