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| 03:27.31 | Cubber | I just installed asteriskNow 2.0.2 64 bit on a server, ran yum update and then tried to enter freepbx from a browser, getting Critical Error retrieve_conf failed, config not applied error and I cannot do anything |
| 03:28.20 | Cubber | if I try to hit apply config it errors with: Error Did not receive valid response from server XHR response code: 200 XHR responseText: undefined jQuery status : parsererror |
| 03:29.46 | Cubber | may have just fixed my issue by adding apache to the asterisk group and vise versa |
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| 03:52.46 | WIMPy | ~freepbx |
| 03:52.46 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 03:52.53 | WIMPy | oh, too late |
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| 08:25.55 | trulsk | we shut down our dialup service yesterday. after 18,5 years. http://www.youtube.com/watch?v=mg-iNiedsf4 |
| 08:29.16 | ChannelZ | I have one of those modems |
| 08:32.32 | F|shie | Can anyone tell me about this: [Jun 2 12:50:28] WARNING[24728]: loader.c:460 load_dynamic_module: Error loading module 'app_intcurl.so': /usr/lib/asterisk/modules/app_intcurl.so: undefined symbol: curl_easy_setopt. Linker issue? |
| 08:33.03 | ChannelZ | probably |
| 08:33.12 | ChannelZ | or if that was from a package, you're missing a library |
| 08:33.29 | F|shie | no i wrote my own module |
| 08:33.46 | F|shie | im using curl library |
| 08:34.34 | ChannelZ | was curl from a package or did you build that? |
| 08:36.00 | F|shie | from a package, curl module which comes with asterisk works okay though |
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| 12:33.28 | mazpe | if i have two interfaces where my devices (ata/phones/softphones) connect too, do i have to specified them in sip.conf ? one interface is VoIP Vlan and the other one is for my external devices. |
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| 12:34.42 | WIMPy | You can;t specify multiple interfaces. |
| 12:34.49 | WIMPy | One or all. |
| 12:35.39 | slav3_kitten | mazpe, are you talking like network interfaces. or like a network interface and network attached fxs? |
| 12:35.56 | mazpe | network interfaces |
| 12:37.08 | mazpe | WIMPy: all works for me… do i just comment it out? |
| 12:39.16 | slav3_kitten | then yea what WIMPy said |
| 12:39.52 | slav3_kitten | mazpe, gimme a second to find the relevant section in the book |
| 12:39.57 | slav3_kitten | btw i suggest getting the book |
| 12:40.14 | mazpe | Yeah, seriously thinking about it |
| 12:41.30 | WIMPy | Set it to 0.0.0.0. Just jeaving it out should do the same. |
| 12:42.11 | slav3_kitten | book says 0.0.0.0 |
| 12:42.12 | mazpe | I'm getting the "no reply to our critical packet" using the softphone |
| 12:42.42 | mazpe | So i'm thinking something within the interface been binded to asterisk |
| 12:42.58 | mazpe | the pbx sits behind a comcast router with an external interface |
| 12:43.12 | slav3_kitten | are the proper ports forwarded? |
| 12:43.15 | mazpe | and an internal interface vlaned for the internal devices |
| 12:43.26 | mazpe | No port forwards.. since it has an external interface |
| 12:43.37 | mazpe | I have even disabled the firewall for testing |
| 12:44.00 | slav3_kitten | well you said behind a router, firewall should only affect internet -> private network an vice versa |
| 12:44.40 | slav3_kitten | with your vlan are both your server and softphone configured to have a virtual interface actually bound to the vlan? |
| 12:45.12 | slav3_kitten | and are you using the correct vlan tagging protocol, also does your router support the transfer of vlan information |
| 12:45.39 | slav3_kitten | like can you ping 1 computer from the other through your vlan'd ip? |
| 12:46.48 | slav3_kitten | example comp one has vlan1 10.1.1.1 and vlan2 11.1.1.1 and comp 2 has vlan1 10.1.1.2 with comp 3 having vlan 2 11.1.1.2 |
| 12:48.54 | mazpe | the soft phones are not on the vlan. |
| 12:49.07 | mazpe | the soft phones are straight from my iPhone acrobits. |
| 12:49.29 | mazpe | everything inside the vlan works fine. |
| 12:50.29 | slav3_kitten | ok hold up a second is the traffic flow... like this |
| 12:50.45 | slav3_kitten | iphone -> internet - |
| 12:50.56 | slav3_kitten | .> router -> vlan -> pbx server? |
| 12:51.47 | mazpe | iphone -> internet -> router -> pbx server |
| 12:52.18 | slav3_kitten | right but your pbx is on a different vlan thus to the router an entirely different network |
| 12:52.47 | slav3_kitten | you need to create the nat and routing rules to tell the router "hey traffic on this port must be tagged with this vlan an sent here |
| 12:55.20 | slav3_kitten | also make sure you let asterisk know you're behind a nat. sip carries IP information |
| 12:55.52 | mazpe | two interfaces… external interface and internal interface is on vlan |
| 12:56.16 | slav3_kitten | wait what? |
| 12:58.54 | slav3_kitten | so everything works with a softphone on your side of the router, like linphone or what not right? |
| 12:59.31 | mazpe | I have two network interfaces on the server… eth0 has an external ip address and eth1/eth1.100 has an internal ip address |
| 13:00.06 | mazpe | the polycom phones connected internally via vlan 100 works fine… but i'm trying to get a soft phone to connect via the external IP address |
| 13:00.59 | slav3_kitten | by external IP you have the server running as your router? |
| 13:03.07 | mazpe | theres a comcast router and we just gave the server an external ip with the comcast router as the gateway |
| 13:05.20 | slav3_kitten | it may be mangling traffic... |
| 13:06.25 | slav3_kitten | if everything works internally you're likely configured properly for asterisk. the voodoo you're doing with dual interfaces instead of just forwarding ports to your asterisk internal address though ... |
| 13:06.39 | slav3_kitten | btw that's not a vlan |
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| 13:17.08 | mazpe | what do you mean its not a vlan? whats not a vlan? |
| 13:17.39 | slav3_kitten | your current setup, vlan tagging is a specification |
| 13:17.58 | slav3_kitten | you just have 2 interfaces one public facing an one internal (horribly insecure btw) |
| 13:19.08 | mazpe | eth1.100 with VLAN=yes is not vlan? |
| 13:19.22 | mazpe | How else do you create a vlan? |
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| 13:20.21 | mazpe | slav3_kitten: the public interface is firewall only allowing sip ports in |
| 13:20.29 | slav3_kitten | on the internal side you have a vlan, but the external side isn't thus your previous "from one vlan to another" is different |
| 13:21.02 | slav3_kitten | are you sure it's allowing the ports correctly. imho it sounds much like a routing issue |
| 13:21.03 | mazpe | oh yeah, externally there is no vlan. |
| 13:21.18 | mazpe | slav3_kitten: for testing i disabled the firewall. |
| 13:22.51 | slav3_kitten | honestly... i can't fathom why you're running it like that. i'd just poke holes in the firewall for the sip phone and have the router route it to the correct vlan thus forgoing any sort of possible loop issues |
| 13:23.41 | WIMPy | What the hack are you discussing there? |
| 13:24.09 | slav3_kitten | WIMPy, the oddest network config i've ever heard of |
| 13:24.13 | mazpe | WIMPy: bit lost. |
| 13:24.29 | WIMPy | What's so special about that? |
| 13:24.39 | mazpe | why is it so odd? |
| 13:24.46 | slav3_kitten | instead of internet->router-> route to vlan all sip traffic -> pbx server |
| 13:25.17 | WIMPy | What difference does the VLAN make? |
| 13:25.28 | slav3_kitten | it's not the difference of the vlan |
| 13:25.52 | mazpe | internet -> router -> (ext if) pbx (int if) <- phone network |
| 13:25.53 | WIMPy | One internal and one external interface. Nothing special about that. |
| 13:26.17 | slav3_kitten | it's the difference of telling a router to send unfiltered traffic to x interface while having y interface on the internal side of the network |
| 13:26.39 | slav3_kitten | vs just forwarding the sip traffic to the server via the router and forgoing the whole dmz thing |
| 13:27.34 | WIMPy | doesn't get it. |
| 13:28.03 | slav3_kitten | in my mind it's a screwy config |
| 13:28.35 | slav3_kitten | i'd much rather have everything behind a router with traffic QoS'd and routed to the proper places via port forwarding an such |
| 13:31.10 | WIMPy | Ok, so now go and hunt for the place where the packets get lost. |
| 13:32.13 | mazpe | and this is what i'm seen on the sip debug |
| 13:32.17 | mazpe | http://nopaste.info/49e347f14b.html |
| 13:36.28 | mazpe | do i have to specify the external ip on sip.conf? |
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| 13:37.42 | slav3_kitten | sorry, not that far into the book |
| 13:38.26 | slav3_kitten | http://www.voip-info.org/wiki/view/Asterisk+SIP+externip |
| 13:38.29 | slav3_kitten | might help |
| 13:40.10 | mazpe | and is it externaddr or externip on 1.8.11.1 |
| 13:40.44 | slav3_kitten | wait you external ip is 1.8.11.1? |
| 13:40.58 | mazpe | asterisk version |
| 13:41.09 | slav3_kitten | oh |
| 13:41.22 | slav3_kitten | you can see how i was a lil confused :P |
| 13:41.45 | slav3_kitten | it's externip = blah |
| 13:41.54 | slav3_kitten | and i think you may need to turn nat on |
| 13:46.11 | WIMPy | You only need externip if your external IP is not the IP of the interface you're using. |
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| 13:46.37 | WIMPy | i.E. if your Asterisk is behind NAT. |
| 13:46.45 | mazpe | WIMPy: where you able to see the log file |
| 13:46.59 | mazpe | better said the debug file |
| 13:47.10 | WIMPy | I hate reading sip debug. |
| 13:47.13 | slav3_kitten | the log makes it look like it makes a connection but terminates it |
| 13:47.26 | mazpe | WIMPy: its not that long :) |
| 13:47.28 | slav3_kitten | is the dialplan setup correctly to handle the call? |
| 13:47.36 | mazpe | its making the call |
| 13:47.50 | mazpe | the call is generated and dropped 20 to 40 seconds into the call |
| 13:48.50 | mazpe | i did noticed it said "Reliably Transmitting (NAT) to 174.25.9.23:8544:" |
| 13:49.03 | mazpe | which is the iPhone ip |
| 13:50.52 | slav3_kitten | i wonder if it's trying to send traffic out of both the router a interface. does your .100 vlan route to the internet at large? |
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| 14:07.43 | [sr] | howdy |
| 14:08.58 | slav3_kitten | *pokes mazpe* unplug your internal vlan'd network cable an see if it works then |
| 14:13.21 | mazpe | i don't see how it will make a diff.. |
| 14:13.46 | slav3_kitten | just try it? |
| 14:14.59 | WIMPy | Hi [sr] |
| 14:15.16 | [sr] | hi WIMPy, how r u? |
| 14:15.37 | WIMPy | mazpe: Looks like someone gables your transmission in between. |
| 14:15.59 | WIMPy | [sr]: Too unproductive. |
| 14:16.15 | WIMPy | mazpe: Looks like only some packets make their way. |
| 14:16.20 | [sr] | WIMPy: same here :) |
| 14:17.23 | WIMPy | mazpe: Or you have excessively extreme delays. |
| 14:18.25 | mazpe | it happens in two different systems same configs |
| 14:18.32 | mazpe | different networks.. but similar setup |
| 14:19.45 | WIMPy | which makes it most likely for the issue to be at the other end. |
| 14:20.24 | slav3_kitten | does your iphone service provider do weird nat stuff? |
| 14:20.33 | mazpe | verizon |
| 14:21.17 | mazpe | gotta run.. bbl |
| 14:21.28 | mazpe | this thing has driven me to huger! |
| 14:21.32 | slav3_kitten | http://danielchoi.net/posts/verizon-blocking-sip-ports/ |
| 14:21.44 | slav3_kitten | it's very possible that verizon is dicking things up |
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| 14:22.25 | mazpe | hmmm.. me try with an AT&T |
| 14:23.45 | slav3_kitten | i'd personally try from like a laptop with linphone from a coffee shop hotspot |
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| 15:52.18 | slav3_kitten | i realize this is likely covered in the book i bought... |
| 15:52.46 | slav3_kitten | but is it possible for a dialplan to detect a stream to an outbound number and park calls till that stream ends? |
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| 16:07.30 | volga629 | What need check to fix it cdr_odbc.c:149 odbc_log: Unable to retrieve database handle. CDR failed. |
| 16:20.13 | slav3_kitten | volga629, i'd think db permissions or existence |
| 16:20.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 16:20.23 | slav3_kitten | but i'm just now learning asterisk |
| 16:20.39 | slav3_kitten | yo [TK]D-Fender |
| 16:20.51 | volga629 | me the same |
| 16:23.21 | slav3_kitten | did you buy the book? |
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| 16:24.01 | slav3_kitten | i highly recommend the book, and building from source since my debian repo version was like 1.6 an they are on 1.8 now |
| 16:24.31 | volga629 | Yes, asterisk 1.8 third edition |
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| 16:25.04 | slav3_kitten | i'd check permissions on the db file |
| 16:25.28 | volga629 | let me see |
| 16:25.45 | volga629 | db file you mean /var/lib/mysql ? |
| 16:25.57 | slav3_kitten | maybe? |
| 16:26.11 | slav3_kitten | i'm not sure where you have the database file being created |
| 16:26.31 | slav3_kitten | my bet would be either it does not exist, or has wrong permissions |
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| 16:27.59 | volga629 | http://fpaste.org/e0me/ |
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| 16:29.53 | slav3_kitten | Error 324 (net::ERR_EMPTY_RESPONSE): Unknown error. |
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| 21:33.21 | ThrobbingPython | hi everyone!!! |
| 21:34.11 | ThrobbingPython | whats the best way to clean up a avaya s8700? |
| 21:34.29 | ThrobbingPython | remove unused ext, vectors and vdn??? |
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| 22:40.54 | ruied | I've changed 'language=pt_m' in sip.conf and copied all files from en to pt_m (including digits and all folders). When I use a language different from en, voicemail stops trying to play for example 1F.alaw In en it is 1.alaw is this a bug? |
| 22:41.48 | WIMPy | That would be a female one. |
| 22:42.49 | Kobaz | i like my 1's to be female |
| 22:44.28 | ruied | ah, ok... but how do make it play just the 1.gsm instead of 1F.gsm |
| 22:44.29 | ruied | ? |
| 22:44.52 | WIMPy | By using a language without genders. |
| 22:45.24 | ruied | so, gender is something like pt_m ? it should be just pt ? |
| 22:46.08 | WIMPy | No, for the samples. |
| 22:46.42 | WIMPy | Some languages only have one version of one, like english. Other languages have different versions of one. |
| 22:51.01 | ruied | I'm Portuguese. There is no language for me, so what I did was copy the en folder to 'pt_m' one, and as needed I'm replacing the en files to portuguese ones ones. But I'm stuck with the 'gender'. Is there any gender option so I can configure? |
| 22:51.54 | WIMPy | I have no idea. They might be hardcoded depending on the language. |
| 22:58.52 | ruied | hmm, maybe... goint to research... |
| 23:00.52 | ruied | there must be some configuration, so if I have to genders I can set it to be reades by voicemail |
| 23:01.01 | ruied | to=two |
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| 23:18.01 | ruied | WIMPy, what if you like Male gender saying 1's digits in voice mail, do you know how to change it ? in en |
| 23:19.43 | WIMPy | In english "one" doesn't have gender. |
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| 23:24.51 | ruied | ok. So it seems to be some kind of asterisk lack of configuration, since if with portuguese there is the possibility to have genders but no way so set it so voicemail can read the correct one. for what I've researched so far there is no set for that... it seems that my solution is to rename all digits file for now or record all of them in portuguese... |
| 23:26.07 | WIMPy | So it's not the correct gender? |
| 23:29.08 | [TK]D-Fender | say.conf <- |
| 23:29.46 | ruied | I would like asterisk to read /var/lib/asterisk/sounds/digits/1.gsm voicemail is trying to read 1F.gsm |
| 23:32.56 | Kobaz | use the source, luke |
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| 23:36.54 | WIMPy | While talking to someone behind NAT, I get some occasional |
| 23:37.00 | WIMPy | WARNING[30319]: res_rtp_asterisk.c:2083 bridge_p2p_rtp_write: RTP Transmission error of packet to 192.168.178.28:5062: Network is unreachable |
| 23:37.19 | WIMPy | but no audible issues. Does that make any sense to anyone? |
| 23:40.49 | WIMPy | Ok. Taking a close look, it happens in a block of 6-7 times every 5 minutes. So it must be the session timer. |
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