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| 04:12.24 | WIMPy | wonders if there's any potential disadvantage in specifying trunk=yes on all iax peers. |
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| 08:12.14 | formater | hi :) |
| 08:13.53 | formater | if I have a complex dialplan, with array, array sorting, etc... all that is now written in php-agi, but the execution time is slow, what do you suggest? Making it in pure asterisk dialplan is not possible... should I create an aserisk module in C ? |
| 08:18.42 | qakhan | hi all, is there any way asterisk make 2 outbound calls when both phones are ringing, asterisk connect both calls and both calle can talk to each other |
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| 08:45.36 | justdave_ | qakhan: yeah, look for "Originate" in the docs |
| 08:47.46 | wdoekes | formater: you should probably look at your php-agi code from a higher level |
| 08:49.10 | wdoekes | and when you say slow, how slow do you mean? |
| 08:49.32 | wdoekes | are there external sources that are slow (large files to be parsed? sql queries to be run?) |
| 08:49.59 | qakhan | justdave_ i know originate can make outbound calls but how to join both calls? |
| 08:50.35 | wdoekes | perhaps some of the work could be cached, or data types/storage rearranged so there is work to be done on each call |
| 08:50.54 | justdave | qakhan: use Local/ with an extension number as your source channel, it'll dial that number first and then place the call when they answer |
| 08:51.14 | justdave | qakhan: I think you have to use the Manager API to do it that way though |
| 08:52.25 | wdoekes | qakhan: or call files |
| 08:52.28 | wdoekes | (res_spool) |
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| 08:54.11 | formater | wdoekes: with php-agi it was slow, so I changed it, to load data using CURL with apache, within extensions.conf, it was too slow, then changed to nginx... and now under heavy load, it takes for nginx/php5-fpm 4 seconds, while the php itself (according to logs) executes very fast |
| 08:54.20 | formater | so, now i want to move away from php |
| 08:54.28 | formater | and rewrite the logic |
| 08:54.45 | formater | the server has like 30-50 calls arriving per second |
| 08:54.58 | formater | and each call executes the php, checking for money, getting route list, etc |
| 08:55.10 | formater | but that part is optimzed, mysql is optimized, etc... but executing the php itself is slow |
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| 08:55.39 | formater | so, that why i wonder what to do. In past the logic was written in extensions.conf, but because it has complex rules, the code was not readable |
| 08:55.59 | formater | that's why me moved to php :) but now, probably i should write an asterisk module, so the execution will be fast |
| 08:56.29 | justdave | is it having to spin up a php interpreter for each call to the php? |
| 08:56.46 | justdave | or does asterisk have some equivalent to apache's mod_php to run it with? |
| 08:56.54 | formater | yes, i believe that initial issue with php-agi was, that it loaded php interpreter all the time with each call |
| 08:57.08 | formater | then i tought, that if i move the php into a web server |
| 08:57.17 | formater | like apache with mod_php |
| 08:57.32 | formater | then it will be faster, because it not leads to load the interpreter and the modules all the time |
| 08:57.50 | formater | but got to a point, where apache was not fast enough (probably i coudln't configure it correctly) |
| 08:57.55 | justdave | and then you still get the fork/spawn overhead trying to shell out to curl probably |
| 08:58.11 | formater | then moved to nginx + php5-fpm, which was the fastest up to know, but again, reached the limits :) |
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| 08:59.07 | formater | i use func_curl i guess, and as I checked asterisk source, it's pretty simple, usign curl library |
| 08:59.13 | formater | so i guess that's not an issue |
| 09:00.52 | formater | now checking if php can be made faster, by using some opcache (like APC or xcache)... but that's not asterisk topic anymore :D So.. do you suggest writing my own asterisk module? Or asterisk dialplan can be written in some other language, which is supported by default by asterisk, so executing is fast ? |
| 09:01.49 | justdave | I'm having a weird issue on my system with one-way audio on incoming IAX calls, but only if the call goes direectly to a phone. If it stops at an IVR menu or voicemail, then audio works fine both directions. |
| 09:02.41 | justdave | a user has a DID number that goes direct to their extension for example... it connects, the remote caller can't hear them. |
| 09:03.05 | justdave | I can take that same DID number and point it at the main phone tree, call it, dial their extension from the IVR, and it works fine |
| 09:03.27 | justdave | the only thing standing out to me is the IVR and the voicemail system both do a Answer() on the channel |
| 09:03.44 | justdave | but shouldn't that part happen as part of the Dial() when it connects it to the phone anyway? |
| 09:04.13 | formater | yes it should |
| 09:04.19 | formater | maybe it's nat issue |
| 09:04.24 | formater | between asterisk and your client ? |
| 09:04.32 | formater | that signaling doesn't go through correctly ? |
| 09:04.49 | formater | have you checked what's going on with tcpdump + wireshark? |
| 09:04.59 | formater | (altough, i have very limited knowledge of IAX protocol) |
| 09:05.11 | justdave | the user's phone is SIP if that matters |
| 09:05.24 | justdave | but it's also local, not NATed |
| 09:05.49 | justdave | and if it was a SIP NAT issue, I'd expect the audio to never work, instead of working when it goes through an IVR |
| 09:06.43 | formater | yeah, but when you have no issue it looks like DID -> Asterisk.. when you have issue, it's DID -> Asterisk -> Client, right ? |
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| 09:09.02 | justdave | formater: yeah, when there's no issue, Asterisk answers first then forwards to the phone |
| 09:09.24 | formater | hm, let me check my notes, i had similar issue |
| 09:09.32 | formater | with an avaya connecting over h323 to asterisk |
| 09:09.37 | formater | and then going out in sip |
| 09:09.56 | justdave | so tcpdump is showing me exactly what I'm observing/listening.... |
| 09:10.10 | justdave | when it goes direct to the phone, there is RTP trafic going only from the phone back to asterisk |
| 09:10.25 | justdave | when it goes through the IVR then forwards to the phone, there is RTP going both directions |
| 09:11.46 | formater | when i had such issue with h323<>sip |
| 09:11.50 | formater | i used the following: http://pastebin.com/UbL63sNU |
| 09:12.07 | formater | notice M(a) in Dial, and macro-a |
| 09:13.18 | formater | i know it looks silly, but it worked :) |
| 09:18.44 | justdave | wouldn't that cause it to go to an answered state even if the user never answers? |
| 09:19.47 | formater | according to: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 09:20.03 | formater | M(x): Executes the macro (x) upon connect of the call (i.e. when the called party answers). |
| 09:20.54 | justdave | according to "core show application Dial" : |
| 09:20.55 | justdave | Execute the specified <macro> for the *called* channel before connecting to the calling channel. |
| 09:22.09 | formater | hm |
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| 09:22.43 | formater | can be :) anyways, if you make a quick test you will see |
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| 09:25.52 | justdave | yeah, just tried it, it does indeed wait until the phone answers to run it |
| 09:26.02 | justdave | but that didn't solve the issue, still one-way audio :| |
| 09:27.03 | formater | :( |
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| 09:28.10 | formater | can you post the tcpdump file, SIP traffic between your asterisk and the client, in both scenarios? when you have audio, and when you don't? |
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| 09:28.34 | justdave | so the macro runs on the channel of the phone |
| 09:28.55 | justdave | when the Answer happens via the IVR menu or voicemail it runs on the calling channel |
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| 10:45.59 | novaflash | hello. i'm using asterisk 11.8 and have figured out a lot of stuff already, but i'm confused about something. i've set up a menu that works when i dial in from an outside line, but won't work when i use an internal extension that i dial from one of my registered sip phones. it does connect, and when i press numbers i can see in the asterisk console that the input is being accepted and that... |
| 10:46.00 | novaflash | ...it is going through the motions of the menu, but the phone is totally silent. again, dialing from outside, everything is fine. i've used for example this setup; https://wiki.asterisk.org/wiki/display/AST/Creating+a+Simple+IVR+Menu - does anyone have an idea of what i am missing? does an internal sip phonecall require a specific extra option to make it 'answer' the phonecall in some way? |
| 10:47.04 | wdoekes | novaflash: did you call Answer()? |
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| 10:53.18 | novaflash | yeah what i did was |
| 10:53.26 | novaflash | exten => 6789,1,Goto(main-menu,s,1) |
| 10:53.28 | novaflash | and |
| 10:53.31 | novaflash | [main-menu] |
| 10:53.32 | novaflash | exten => s,1,Answer() |
| 10:53.34 | novaflash | <PROTECTED> |
| 10:53.35 | novaflash | <PROTECTED> |
| 10:53.42 | novaflash | and other stuff beyond this to handle the selections |
| 10:54.23 | novaflash | so from my sip phone i dial 6789 and it does say it is connected, but the phone is silent. i don't hear 'press 1 or press 2'. when i dial from outside and goto(main-menu,s,1) it works great. |
| 10:58.06 | novaflash | in the console i see it executing 6789@default:1 and goto main-menu, and i see it playing the sounds and waiting for input. when i press a number, it moves on to the next part of the main menu, and so on, but never a sound on the horn or the speaker |
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| 11:06.16 | Chainsaw | novaflash: Could be something as mundane as a NAT problem causing you to lose the audio streams. |
| 11:06.36 | Chainsaw | novaflash: Make a simpler extension that's just an echo test, and fiddle with the settings until you get the echo test working. |
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| 11:07.03 | Chainsaw | novaflash: Only then return to your IVR/menu systems. Troubleshoot one aspect at a time. You're testing two or three here. |
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| 11:12.36 | novaflash | yeah godo point, let's take a few steps back. time to find out how to set up an echo test thing |
| 11:16.43 | novaflash | okay well |
| 11:16.51 | novaflash | how is it possible that even an echo test doesn't work |
| 11:16.56 | novaflash | but my phone works fine in every other situation? |
| 11:23.47 | novaflash | :q |
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| 13:29.04 | marceloamorim | anyone knows if is possible debug the iax2 ? I`m try a new iax but I couldn`t register yet |
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| 13:40.52 | marceloamorim | I found my mistake, with the host=dynamic wasn`t work, now I set host=ip_other_server and worked |
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| 13:59.04 | fforp | marceloamorim: if you want to use 'host = dynamic' you must use 'register => user:passwd@host' |
| 14:00.36 | marceloamorim | I did, but wasn`t work, but don`t worry, work with specific host |
| 14:01.18 | marceloamorim | my problem now is being make a call |
| 14:01.26 | marceloamorim | Unable to create channel of type 'IAX2' (cause 20 - Subscriber absent) |
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| 14:06.08 | marceloamorim | chan_iax2.c:4676 __auto_congest: Auto-congesting call due to slow response |
| 14:07.15 | marceloamorim | if I qualify the iax, he`s showing unreachable |
| 14:07.50 | marceloamorim | don`t make sense, the icmp response with 0.150ms |
| 14:08.03 | marceloamorim | and don`t lose any packet |
| 14:08.12 | WIMPy | Firewall? |
| 14:08.43 | WIMPy | Although slow response sounds more like packet loss or a laggy application. |
| 14:09.00 | marceloamorim | don`t have any, my iptables just block some bot networks |
| 14:10.03 | marceloamorim | and other machine is ipfw rule > 65535 allow ip from any to any |
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| 14:19.28 | fforp | here is my config http://pastebin.com/4g7yTHiN |
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| 14:19.56 | fforp | just commen 'register' and replace 'host=' |
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| 14:23.54 | marceloamorim | oh man, thx I`ll try with your config |
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| 14:26.22 | gaps | is there any solution to integrate with payment gateways? |
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| 14:44.46 | marceloamorim | fforp: still unreachable =( |
| 14:44.52 | marceloamorim | don`t make sense |
| 14:45.48 | marceloamorim | with qualify I can see that iax is offline |
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| 14:59.25 | Katty | you mean doesn't make sense. |
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| 15:03.05 | marceloamorim | yeah, my englis is bad, my main language is portuguese, but I guess you don`t speak right? sorry about that mistake =) |
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| 15:06.14 | jaflong | Hi Everyone. |
| 15:06.25 | jaflong | I am getting this error WARNING[31977][C-00000009]: chan_sip.c:10657 process_sdp: Can't provide secure audio requested in SDP offer |
| 15:06.38 | jaflong | From the sdp http://pastebin.com/J1vSFh2w can anyone suggest why secure audio is not provided |
| 15:07.39 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 15:11.22 | marceloamorim | its weird, I change the IP of my second server and change the register user:password@newip and when I reload module and reload iax2, after that I use iax2 show peers and the old ip still there |
| 15:11.59 | WIMPy | Sure |
| 15:12.28 | WIMPy | 'iax2 show peer' shows your peers. To see registrations, use 'iax2 show registry'. |
| 15:14.14 | marceloamorim | but the peer ip was changed |
| 15:14.35 | WIMPy | That's not what you said. |
| 15:16.59 | marceloamorim | my B server is peer of my A server, so I change the ip from b server |
| 15:17.26 | *** join/#asterisk loggiew (~logan@node2.thiscluster.com) |
| 15:17.27 | marceloamorim | but its ok, my english bad anyway |
| 15:17.59 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 15:18.49 | loggiew | both of my devices are behind a NAT here at home connecting to my asterisk server on a public IP. They both appear to be looking for a response on the external port 5060 of my local NAT'ed IP |
| 15:18.56 | loggiew | am I misunderstanding something? |
| 15:19.25 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 15:19.54 | dan_j | Hi, is anyone aware of any SIP blocking in India by ISPs? |
| 15:20.48 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 15:21.40 | marceloamorim | apparently reload iax2 modules isn`t work appropriately, so I restart the asterisk and seems it worked |
| 15:38.30 | *** join/#asterisk DevWork__ (~DevWork@74.10.31.134) |
| 15:38.47 | DevWork__ | Is it normal to route RTP through the asterisk box from an ip phone to the destination? |
| 15:41.28 | devlap | I am calling an outside number, and I expected my phones to send their voice to the asterisk box, and have it route out from there. |
| 15:41.51 | devlap | It seems the phone is trying to communicate out directly to our sip provider when it does RTP |
| 15:41.57 | WIMPy | devlap: It's the version that's more likely to work. |
| 15:42.21 | WIMPy | And that's the more efficient way. |
| 15:42.23 | *** join/#asterisk mokmeister (~mokmeiste@86.41.108.76) |
| 15:43.56 | devlap | How do I get it to not, uhh I guess thats the sip reinvite? |
| 15:44.03 | devlap | For outbound calls. |
| 15:44.06 | WIMPy | yes |
| 15:44.33 | devlap | I don't want the phones punching through the firewall directly, especially when SIP requires such a wide swath of ports, |
| 15:44.49 | WIMPy | SIP uses one port. |
| 15:44.54 | devlap | sorry RTP |
| 15:45.18 | WIMPy | And RTP should work atumatically without explicitely opening ports. |
| 15:45.28 | devlap | We default block all. |
| 15:45.41 | devlap | so unless its specifically allowed, the traffic is dropped. |
| 15:45.52 | WIMPy | everyone does. |
| 15:46.29 | devlap | Ok well right now, its getting blocked outbound RTP, because its coming from the phones, not the asterisk box. |
| 15:46.30 | WIMPy | But there's an outgoing stream. And it can be "related" to the SIP traffic. |
| 15:47.33 | WIMPy | Oh, you block all outgoing as well? |
| 15:47.53 | devlap | We default block all, yes. |
| 15:49.56 | devlap | ill try canreinvite = no on my sip outbound. |
| 15:50.54 | WIMPy | canreinvite, directmedia, directrtpsetup, depending on your version. |
| 15:51.21 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 15:51.56 | devlap | I have directmedia=no set on it already. |
| 15:52.26 | devlap | asterisk 12.1.0 |
| 15:53.49 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 15:55.38 | *** join/#asterisk newtonr (~newtonr@173-17-135-67.client.mchsi.com) |
| 15:55.38 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:05.09 | devlap | I did sip show on my outbound peer and I see directmedia=no in its config so its loading that. |
| 16:08.36 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 16:16.20 | *** join/#asterisk hellc2 (~elio@100.Red-88-26-250.staticIP.rima-tde.net) |
| 16:16.58 | hellc2 | Hi :) |
| 16:17.10 | devlap | Hmm. What about ringing two seperate asterisk installs. My sip provider can send calls down too both installs, am I fine until I do an Answer() or Progress(), meaning will both sides be able to pickup until that operation? |
| 16:21.40 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 16:22.53 | devlap | I don't know why its doing it, but I can't make it not reinvite. |
| 16:23.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 16:25.19 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 16:25.20 | gaps | is there any solution to integrate with payment gateways? |
| 16:25.45 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 16:26.01 | devlap | http://pastebin.ca/2679759 is my sip.conf |
| 16:26.10 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 16:27.17 | hellc2 | gaps, payment gateways? Asterisk? |
| 16:27.28 | devlap | Its still sending rtp from my client to our sip provider directly. |
| 16:34.24 | _Corey_ | gaps: Define 'integrate' |
| 16:35.29 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 16:36.33 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:37.45 | gaps | hellc2: yes to integrate with asterisk |
| 16:38.32 | hellc2 | gaps, something like a2billing? |
| 16:40.41 | *** join/#asterisk alami (~ialami@unaffiliated/alami) |
| 16:41.32 | gaps | hellc2: not a2billing.. from asterisk if i dial a number it will pass on to payment gateway system and then as a caller i can enter my credit card number |
| 16:41.43 | gaps | hellc2: and so on |
| 16:42.24 | devlap | Straight up asterisk? You can build an IVR system sure. |
| 16:42.42 | devlap | http://codezone.apstel.com/tiki-view_forum_thread.php?topics_offset=1&forumId=3&comments_parentId=472 |
| 16:43.40 | [TK]D-Fender | gaps: It's your system... set up your dialplan however you want. Integrate whatever you want. |
| 16:45.01 | gaps | [TK]D-Fender: agree... but payment gateway we have to interact with banking system's and all if am not wrong... |
| 16:45.25 | [TK]D-Fender | gaps: Go look at what the bank offers an an API <- |
| 16:45.26 | devlap | gaps: payment providers provide their own apis, you have to interface with them, it involves some coding, yea. |
| 16:45.27 | alami | [TK]D-Fender : i have a server for a friend with elastix and i use only asterisk configuration files and command line, if make change for exemple at extensions.conf, will that be ok or i have to made all changes via elastix? |
| 16:45.59 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 16:46.02 | cj | hey folks |
| 16:46.08 | [TK]D-Fender | gaps: Don't ask what Asterisk can do... you have access to an entire operating system behind it... ask what your SERVICE offers as an API... and the realize there is pretty much certainly some programming to do to use it |
| 16:47.09 | cj | if I want either 12062265809 or 2062265809 to Dial(IAX2/foo/${EXTEN},30), what do I use for the extension string? |
| 16:47.51 | novaflash | sorry to bother you people with this simple question, but perhaps someone here knows; what could be the explanation behind a phone not being able to do an echo test, but the phone works fine on the same server for everything else? i'm asking because i want to create a menu system and test it internally. |
| 16:49.09 | cj | novaflash: have you set debug and verbose to 99 from the console? |
| 16:49.14 | hrnt | novaflash: so echo test does not work, but everything else does? (so you can hear voice from asterisk when doing other stuff) |
| 16:50.11 | novaflash | hrnt: i use this phone daily to receive calls through asterisk and didlogic, and to make calls, through asterisk and didlogic. playback and background are silent too. the console shows the sounds are being played to the phone, but there is no sound on the phone. |
| 16:50.16 | [TK]D-Fender | [12:47]cjif I want either 12062265809 or 2062265809 to Dial(IAX2/foo/${EXTEN},30), what do I use for the extension string? <- just put the number |
| 16:50.23 | [TK]D-Fender | cj: no need for variables there |
| 16:50.44 | [TK]D-Fender | novaflash: pastebin your echo test |
| 16:50.46 | cj | [TK]D-Fender: but I don't want two extension lines. I just want one... |
| 16:50.46 | [TK]D-Fender | ~pb |
| 16:50.47 | 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. |
| 16:50.49 | [TK]D-Fender | ^^^^ |
| 16:50.49 | novaflash | cj: no but i have run asterisk -rvvvv and kept an eye on the asterisk console as i made the internal call. it shows things starting and doing stuff, but the phone remains silent. it says it is connected |
| 16:50.59 | novaflash | [TK]D-Fender: thanks, i know what pastebin is ;) |
| 16:51.03 | alami | ~elastix |
| 16:51.03 | infobot | extra, extra, read all about it, elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 16:51.53 | [TK]D-Fender | cj: If its patterns you want to save on.. then just make a pattern that matches both. |
| 16:51.58 | hrnt | novaflash: also, when looking at asterisk console, does Playback terminate immediately and go directly to Echo? |
| 16:52.23 | cj | [TK]D-Fender: yes, can you tell me where to find the pattern format? |
| 16:52.31 | cj | looks for his PDF |
| 16:52.59 | novaflash | hrnt: no it takes time for it to finish the playback and then moves on. |
| 16:53.08 | [TK]D-Fender | cj: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics.html |
| 16:54.08 | novaflash | [TK]D-Fender: i have pastebinned the output when i call my internal extension 600 from my sip phone. http://paste.kde.org/pl5xtyzpg/qf8zrg/raw |
| 16:55.53 | pabelanger | anybody use avconv to transcode into sln? |
| 16:56.21 | cj | found the section in the PDF, but I'm looking for something that matches 1 or nothing |
| 17:01.17 | [TK]D-Fender | cj: Actually, I just missed something in your 2 numbers... you can't do a "one or nothing". That'll require 2 patterns |
| 17:01.36 | *** join/#asterisk lambda79 (~lamdb@pom51-3-88-165-80-74.fbx.proxad.net) |
| 17:01.47 | [TK]D-Fender | novaflash: Do you hear the recording? |
| 17:01.48 | devlap | [1]! |
| 17:01.53 | cj | thank you devlap! |
| 17:01.57 | devlap | matches zero or more 1's |
| 17:02.16 | cj | how about 1{0,1} ? ;-) |
| 17:02.16 | [TK]D-Fender | no.... |
| 17:02.17 | cj | or 1? |
| 17:02.29 | [TK]D-Fender | [13:01]devlap[1]! <- incorrect |
| 17:02.50 | novaflash | [TK]D-Fender: nope. completely silent. interestingly, when i do the echo test, playback, etc.. but i let it come in from an external phoneline, it works fine. it's only internally that it is dead silent. but it goes through the motions, so to speak. |
| 17:02.50 | pabelanger | avconv -i "http://176.9.18.27:8000" -ar 8000 -ac 1 -acodec pcm_mulaw test.ul gets me ulaw |
| 17:02.54 | pabelanger | but looking for sln |
| 17:03.04 | [TK]D-Fender | cj: TWO patterns required for this. |
| 17:03.45 | devlap | oh yea it is any character the ! |
| 17:03.49 | devlap | not just a modifier. |
| 17:03.53 | [TK]D-Fender | novaflash: "sip set debug on" |
| 17:04.02 | [TK]D-Fender | devlap: That's the first part of your error |
| 17:04.10 | cj | how about a rule for all extensions starting with 1 go on to check whether the pattern matches NXXNXXXXXX ? |
| 17:04.23 | [TK]D-Fender | devlap: then there is the fact you said "zero or more ONES". It has no relationship to "ones" |
| 17:04.47 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 17:04.48 | [TK]D-Fender | cj: that is already a second pattern.. what you DO in your dialplan is another matter. |
| 17:04.53 | [TK]D-Fender | cj: that IS a second pattern. |
| 17:05.08 | [TK]D-Fender | novaflash: PB another full call. |
| 17:05.30 | novaflash | [TK]D-Fender: do you mind if i send the pastebinned link to you privately as it contains all my info? |
| 17:05.41 | [TK]D-Fender | novaflash: fine |
| 17:06.47 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:2876:3269:1e31:eed3) |
| 17:08.40 | novaflash | alright i've done just that, take your time ;) |
| 17:09.03 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 17:10.17 | devlap | As I said, the ! matches zero or more characters, it is not a modifier to the previous reference. |
| 17:10.38 | devlap | I'd just do it with two patterns :) |
| 17:10.44 | devlap | Makes it much more clear the intent. |
| 17:11.17 | [TK]D-Fender | [13:01]devlapmatches zero or more 1's <- wasn't what you said |
| 17:11.42 | devlap | Anda fter that, yea it is any character the !, not just a modifier. |
| 17:12.31 | devlap | Like regex, there are character selection syntax, similar with the brackets, you can follow those with selection quantifiers, which act as a modifier to the previously defined character set. |
| 17:13.10 | devlap | Speaking of, can you do regex in a dialplan? |
| 17:13.17 | [TK]D-Fender | novaflash: make sure you have the ports in rtp.conf forwarded/allowed inbound on your server and for it to send to any as well. |
| 17:13.26 | [TK]D-Fender | devlap: not as the pattern match |
| 17:13.29 | devlap | Without making a custom application or something, out of the box. |
| 17:13.57 | devlap | Ah. Probably be much more trouble than its worth. |
| 17:13.57 | [TK]D-Fender | devlap: there is func_regex for expression matching, but not the pdial-pattern itself |
| 17:14.08 | devlap | A simple dialplan is a good dialplan. |
| 17:14.09 | novaflash | [TK]D-Fender: okay i'll just flush iptables out momentarily |
| 17:14.22 | [TK]D-Fender | novaflash: you need 5060, and whatever is in rtp.conf |
| 17:15.41 | novaflash | all ports are open. iptables flushed. default policies on the chain to ACCEPT. it's on a public IP straight to the internet, no intermediate firewalls. no joy. |
| 17:16.14 | novaflash | rtpstart=34000,rtpend=36000 |
| 17:16.19 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 17:16.24 | novaflash | too narrow? |
| 17:16.49 | devlap | [TK]D-Fender, what about a hardphone communicating directly with the sip trunk for RTP when canreinvite=no and directmedia=no on the sip peer is set? my sip.conf http://pastebin.com/zWSyrV4B |
| 17:18.16 | [TK]D-Fender | devlap: this is not his scenario. |
| 17:18.37 | [TK]D-Fender | devlap: and you still need the ports if you epect * to make any sound ever... like I dunno... prompts, voicemail, or anything at all... |
| 17:18.45 | devlap | Thats my scenario. |
| 17:19.16 | [TK]D-Fender | devlap: If all you expec t is * to act as a call-relay and never touch audio ever... then maybe. But that's still not particularly good... |
| 17:19.31 | [TK]D-Fender | devlap: that means no progress, no recording, no queueing, no MoH, NOTHING. |
| 17:19.32 | devlap | I want it to touch audio. |
| 17:19.38 | [TK]D-Fender | devlap: at which point.. why have Asterisk? |
| 17:19.39 | devlap | Its bypassing. |
| 17:20.02 | devlap | Hence my problem |
| 17:20.24 | [TK]D-Fender | devlap: You are not very clear what your problem is. Start over |
| 17:21.01 | devlap | Alright. I make a call from my phone. when the other side picks up, the phone starts sending rtp data to my sip trunk directly. |
| 17:21.12 | devlap | I have canreinvite and directmedia=no as shown in my sip.conf link. |
| 17:21.23 | [TK]D-Fender | devlap: Show the call. |
| 17:21.30 | *** part/#asterisk lambda79 (~lamdb@pom51-3-88-165-80-74.fbx.proxad.net) |
| 17:21.32 | devlap | sip set debug on good? |
| 17:21.59 | [TK]D-Fender | and "core set verbose 10" |
| 17:22.02 | devlap | thats flooded with peering message. |
| 17:22.47 | devlap | How can I get them to log to a file instead of the console? |
| 17:23.35 | devlap | just do a console direct sec. |
| 17:31.00 | devlap | https://db.tt/S9DyclDr is the log |
| 17:35.25 | [TK]D-Fender | devlap: [KSNCA00-PH00*CLI> [0KReliably Transmitting (no NAT) to 38.96.36.14:5060: Contact: <sip:7144348010@10.42.0.20:5060> |
| 17:35.48 | [TK]D-Fender | devlap: Your server is not setup right to work from behind NAT, and you are handing them your server's PRIVATE address for return contact. |
| 17:35.58 | [TK]D-Fender | devlap: Very first part is starting wrong |
| 17:36.44 | devlap | The peer is as they said to set it up. Being that I know nothing of their internal hosts, I just get a port with dhcp on it, and it gives me their internal address. |
| 17:36.53 | devlap | Ill swap it over. |
| 17:37.00 | [TK]D-Fender | devlap: Your core setup is wrong. |
| 17:37.07 | devlap | how is that? |
| 17:37.10 | [TK]D-Fender | devlap: I didn't say the peer to them was |
| 17:37.29 | [TK]D-Fender | devlap: YOU are giving them the wrong IP because you didn't tell * how to get your WAN IP |
| 17:37.50 | devlap | It depends on the trunk. |
| 17:38.06 | [TK]D-Fender | no |
| 17:38.11 | devlap | It doesnt? |
| 17:38.19 | [TK]D-Fender | no, YOUR IP is YOUR IP |
| 17:38.23 | devlap | We have several. |
| 17:38.30 | [TK]D-Fender | devlap: * doesn't multi-home |
| 17:38.37 | devlap | That provider, gives us a dynamic ip, and we route that over that trunk. |
| 17:38.39 | [TK]D-Fender | Your server should have one |
| 17:38.54 | devlap | ok well I can get it to one then, its a dynamic ip all the same though. |
| 17:39.09 | workingcats | it doesnt matter if its dynamic, is it a routable IP? |
| 17:39.11 | *** join/#asterisk AlHafoudh (~textual@dsl-static-173.212-5-200.telecom.sk) |
| 17:39.14 | [TK]D-Fender | Doesn't matter if it's dynamic.. what matters is that * KNOWS what it is |
| 17:39.15 | AlHafoudh | hi |
| 17:39.19 | devlap | Uhh routable to our sip trunk provider, yes. |
| 17:39.27 | AlHafoudh | is it possible to create users type=peer in asterisk web gui? |
| 17:39.43 | [TK]D-Fender | AlHafoudh: What "web gui"? |
| 17:40.07 | AlHafoudh | the digium branded blue one, i dont know how it is called :( |
| 17:40.10 | devlap | I set externaddr, it was previously, will see how that goes. |
| 17:40.25 | [TK]D-Fender | [13:40]AlHafoudhthe digium branded blue one, i dont know how it is called :( <- Switchvox? |
| 17:40.35 | [TK]D-Fender | AlHafoudh: because the only other Digium one... is orange |
| 17:41.12 | AlHafoudh | [TK]D-Fender: its asterisk... |
| 17:41.40 | [TK]D-Fender | AlHafoudh: picpaste.com <- screenshot it |
| 17:41.42 | AlHafoudh | yup https://dl.dropboxusercontent.com/u/350675/digium.png |
| 17:42.07 | [TK]D-Fender | AlHafoudh: That looks liek AsteriskGUI and has been dead for YEARS and is not supported |
| 17:42.30 | AlHafoudh | I thought so :( i got it in my synology NAS as an appliance package |
| 17:42.51 | [TK]D-Fender | I keep seeing people coming in here about those stupid toasters..... |
| 17:43.09 | [TK]D-Fender | Anyway... that GUI has no support anymore. |
| 17:43.16 | AlHafoudh | :) actually the synology nas is quite good... |
| 17:43.30 | Qwell | It's a NAS. Network Attached Storage. |
| 17:43.32 | [TK]D-Fender | AlHafoudh: As a NAS.... probably. |
| 17:43.35 | AlHafoudh | i see, any idea how to do it? |
| 17:43.36 | Qwell | Not PBX. |
| 17:43.44 | AlHafoudh | @Qwell: yes |
| 17:44.02 | Qwell | Stop trying to use your storage device as something other than a storage device. |
| 17:44.29 | novaflash | Qwell: a fridge with facebook. that's what everybody wants ;) |
| 17:44.30 | [TK]D-Fender | Or at the very least, configure * yourself and not via an abandoned GUI we cannot support |
| 17:44.38 | Qwell | novaflash: sadly, you aren't even wrong. |
| 17:44.45 | novaflash | heh |
| 17:45.52 | devlap | https://db.tt/S9DyclDr updated with a new log. I set externaddr |
| 17:46.07 | AlHafoudh | i was just asking if anyone knows how to do it with this old crap, ok, thanks for help |
| 17:46.56 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 17:50.49 | [TK]D-Fender | devlap: [KSNCA00-PH00*CLI> [0KReliably Transmitting (NAT) to 38.96.36.14:5060: INVITE sip:17147240410@sws0.wlphone.net SIP/2.0 <- why is the NAT now? |
| 17:51.08 | devlap | Why? |
| 17:51.20 | devlap | all I did was set externaddr and reload |
| 17:52.18 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 17:52.31 | [TK]D-Fender | Go check your peer. It is not right |
| 17:54.22 | devlap | http://pastebin.com/761gP5iv is what its set to atm |
| 17:59.26 | [TK]D-Fender | devlap: I now see the IP handed to your phone as that of *, not the external peer |
| 17:59.35 | [TK]D-Fender | devlap: should have audio going through * at this point |
| 18:00.01 | [TK]D-Fender | <--- Reliably Transmitting (no NAT) to 10.42.3.101:5060 ---> SIP/2.0 200 OK o=root 427336507 427336507 IN IP4 10.42.0.20 |
| 18:00.14 | devlap | let me place a call. |
| 18:00.28 | [TK]D-Fender | isn't that what you gave me? |
| 18:01.07 | [TK]D-Fender | Hhm... something odd below... |
| 18:01.08 | devlap | It is. |
| 18:01.11 | devlap | But I don't see that. |
| 18:01.21 | *** join/#asterisk salz212 (~chatzilla@182.178.154.51) |
| 18:01.36 | [TK]D-Fender | [KSNCA00-PH00*CLI> [0KReliably Transmitting (NAT) to 38.96.36.14:5060: INVITE sip:17147240410@38.96.36.14 SIP/2.0 X-asterisk-Info: SIP re-invite (External RTP bridge) |
| 18:01.36 | devlap | I see the phone sending out what looks like RTP to 38.96.36.14 |
| 18:01.44 | [TK]D-Fender | LOOKS like a reinvite. |
| 18:01.56 | [TK]D-Fender | Which it shouldn't do based on * telling your phone to go local/. |
| 18:02.33 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 18:02.43 | [TK]D-Fender | Something is whacked here... |
| 18:03.00 | [TK]D-Fender | because The config shows one thing... and the debug another.,...\ |
| 18:03.11 | devlap | Ok well I can restart asterisk and give a fresh one. |
| 18:03.36 | [TK]D-Fender | check your settigns and give it a fresh start |
| 18:04.09 | smkelly | stares at file |
| 18:04.18 | file | smkelly, go away! |
| 18:04.24 | smkelly | pouts |
| 18:04.50 | smkelly | leaves file to his death by snow burial |
| 18:08.41 | devlap | https://db.tt/S9DyclDr Reset the box. Set sip debug on. verbose 10. placed a call with the posted settings as current. |
| 18:11.18 | salz212 | hi all, I have a scenario UAC1-> Asterisk 1-> Asterisk2 -> Asterisk1 -> UAC2 and I have MOH confirgured to be a non default one from dial-plan, no matter what I do it doesn no play MOH on asterisk1 as asterisk2 does not have moh loaded. |
| 18:12.12 | devlap | What I see in my firewall is the phone .4.101 hitting my firewall source port udp 5004 -> 38.96.36.14 |
| 18:12.39 | devlap | .3.101 my mistake. |
| 18:13.31 | salz212 | what does Asterisk need for MOH to play? a re-INVITE as per RFC is coming from the enpoint but it seem having 2 Asterisk servers is messing it up. |
| 18:15.15 | *** join/#asterisk imcdona (~Thunderbi@75-151-101-100-Washington.hfc.comcastbusiness.net) |
| 18:15.45 | *** join/#asterisk danjenkins_ (~dan@80.4.22.208) |
| 18:18.33 | *** join/#asterisk nny (~Scott@174.107.218.2) |
| 18:19.32 | nny | silly question: If I change a user name in sip/voicemail.conf etc. but they say the email message for sending voicemails as attachments still shows the old user (and voicemail show users is correct), where do I look/what's the best way to fix it? |
| 18:19.55 | nny | I assume "restarting asterisk" would work but seems heavy haned and just a knee jerk |
| 18:20.23 | marceloamorim | fforp: are you there? |
| 18:21.50 | *** join/#asterisk DiscusZ (~DiscusZ@S0106106f3fe79fa4.cg.shawcable.net) |
| 18:22.23 | [TK]D-Fender | [KSNCA00-PH00*CLI> [0KTransmitting (no NAT) to 38.96.36.14:5060: <- back to non-nat at least.... |
| 18:23.12 | *** join/#asterisk napnap (~napnap@95-141-110-114.as16211.net) |
| 18:23.15 | *** join/#asterisk vostro (~voix@cpe-108-182-99-183.nyc.res.rr.com) |
| 18:23.39 | nny | or a more direct question where does the voicemail app get the name of the sending and receiving user when sending an email? Is it just the channel CID variable? |
| 18:23.50 | devlap | [TK]D-Fender, Yea I see that. |
| 18:23.51 | DiscusZ | anyone around to answer questions on how to get asterisknow to detect the DID (I have 2 SIP numbers and would like to route them to 2 different IVRs) but even the cdr logs do not show anything inthe DID column |
| 18:24.06 | vostro | does any know what hardware i need to connect analog phone lines to a server running asterisk? Can I use these Digium cards? http://www.digium.com/sites/digium/files/content/image/digium-a4b-analog-card.png |
| 18:24.44 | napnap | Hi all...I use an openvox B400P card, dahdi recognize it well but not asterisk...dahdi_scan (active yes, alarms=OK), in asterisk console dahdi show channels return nothing... any suggestion ? |
| 18:24.47 | vostro | im trying to connect to the phone lines themselves, not handsets |
| 18:24.48 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 18:26.14 | [TK]D-Fender | [14:24]napnapHi all...I use an openvox B400P card, dahdi recognize it well but not asterisk...dahdi_scan (active yes, alarms=OK), in asterisk console dahdi show channels return nothing... any suggestion ? <- that first check does not tell us chan_dahdi is configured for that card or that dahdi is initialized |
| 18:28.15 | *** join/#asterisk LemensTS (~t15@8.33.19.98) |
| 18:29.35 | LemensTS | Hello I have 2 polycom 321's offsite connected to asterisk in the cloud. When they call each other and hangup, it is not hanging up the channel in asterisk. Any idea what to look at? Sip debug didn't show a BYE packet |
| 18:30.02 | LemensTS | It might show a BYE packet if I wait a long time possibly...didn't try |
| 18:34.16 | [TK]D-Fender | vostro: cARDS LIKE THAT, YES |
| 18:34.27 | vostro | ok thanks |
| 18:34.30 | [TK]D-Fender | devlap: PB your current config... |
| 18:34.43 | [TK]D-Fender | dev previous didn't match... want to see what you supposedly have now... |
| 18:36.17 | devlap | Which config? SIP? |
| 18:37.30 | devlap | http://pastebin.com/dqA3MTY8 |
| 18:37.39 | devlap | just sftp'd it off |
| 18:42.34 | *** join/#asterisk imcdona (~Thunderbi@75-151-101-100-Washington.hfc.comcastbusiness.net) |
| 18:43.53 | [TK]D-Fender | devlap: Ok, this is quite odd. |
| 18:44.06 | devlap | I agree it is odd. |
| 18:45.09 | napnap | (my dahdi-channels.conf is empty... I don't know why dahdi_genconf don't regenerate a good one) |
| 18:45.39 | [TK]D-Fender | Hey dev's here -- Looks like * 12.1.0 is not respecting disabling re-invites (directmedia=no) --- Debug : https://db.tt/S9DyclDr --- Config: http://pastebin.com/dqA3MTY8 |
| 18:45.55 | [TK]D-Fender | Cound someone take a look and see if there is something we've missed... or help ID as an actual bug. |
| 18:46.28 | [TK]D-Fender | Could* |
| 18:49.52 | *** join/#asterisk marceloamorim (bd5ac048@gateway/web/freenode/ip.189.90.192.72) |
| 18:52.33 | file | huh |
| 18:53.24 | [TK]D-Fender | Reliably Transmitting (no NAT) to 38.96.36.14:5060: INVITE sip:17147240410@38.96.36.14 SIP/2.0 --- X-asterisk-Info: SIP re-invite (External RTP bridge) |
| 18:53.36 | file | does indeed look like it |
| 18:53.42 | [TK]D-Fender | seems to be doing a reinvite even though his config says "no" |
| 18:53.53 | devlap | Yea, makes me sad :( |
| 18:54.03 | [TK]D-Fender | devlap: but at least you aren't insane! |
| 18:54.10 | [TK]D-Fender | Well... ok, you aren't WRONG about this that is |
| 18:54.21 | *** join/#asterisk chazzam (~chazz@donutokyo.info) |
| 18:54.25 | devlap | Well my config as you saw it first, was the product of me trying to make it work. It was originally about how we have it now. |
| 18:54.48 | devlap | I had externaddr ( my first post of it, has a ; infront of it ) that was after tweaking. |
| 18:54.49 | *** join/#asterisk thafreak (~thafreak@unaffiliated/thafreak) |
| 18:54.59 | [TK]D-Fender | devlap: well the first run didn't match at all... this... just doesn't match on one thing... that is at least consistent |
| 18:55.01 | devlap | Now we have come full circle, and im still getting it. |
| 18:55.05 | thafreak | so what's everyone's favorite handset these days? :) |
| 18:55.24 | [TK]D-Fender | Polycom > All |
| 18:55.28 | paulc | thafreak: I always liked Polycoms, but find their VVX touch screens a bit wonky sometimes |
| 18:55.42 | devlap | I am using polycoms for conference. GXP wasn't bad for the price. |
| 18:55.48 | file | file an issue at https://issues.asterisk.org/jira and newtonr will reproduce, further isolate probably, and then we can fix it |
| 18:55.58 | paulc | Cisco SPA508G worked well at the current place for the PBX replacement project |
| 18:56.11 | file | re: the reinvite |
| 18:56.11 | paulc | GXPs are nice enough but I find their usability a bit wonky in places |
| 18:56.19 | thafreak | i'm getting nothing but complaints from customers with the grand streams...so looking for a new brand |
| 18:56.29 | paulc | thafreak: what are their issues? |
| 18:56.32 | devlap | thafreak, what complains, which brand, I just bought 50. |
| 18:56.36 | [TK]D-Fender | thafreak: Where are you located? |
| 18:56.37 | devlap | *model |
| 18:56.58 | thafreak | i think they are gxp 1200 |
| 18:57.15 | devlap | ahh im using 2100/2160s. |
| 18:57.20 | devlap | Can't comment on those. |
| 18:57.51 | thafreak | call quality...various complaints...other side doesn't hear anything wrong, it's just my clients side... |
| 18:58.09 | devlap | Sounds like a firewall or nat timeout issue. |
| 18:58.26 | thafreak | no firewall or nat |
| 18:58.35 | thafreak | phones on the same lan as asterisk |
| 18:59.18 | thafreak | calls going out over pots lines |
| 19:00.04 | thafreak | i don't even think it's the handsets per-se, but i thin it has something to do with their environment |
| 19:00.09 | devlap | interesting. I got an old school meridian pbx guy running my cat 6. He said the call quality is amazing via our gxp's and asterisk. He would've never guessed it was VOIP. |
| 19:00.37 | [TK]D-Fender | VoIP is not an inherent quality issue |
| 19:00.57 | thafreak | i have one at my house, and i call over the internet to their phone system and then make calls out over their pots, and it's fine for me |
| 19:01.05 | [TK]D-Fender | bad audio is either bad at the PSTN-side, crappy handsets, or serious networking issues |
| 19:01.39 | thafreak | i have a feeling it's something to do with their environment...fluorescent lighting emf or something |
| 19:02.17 | thafreak | last complaint was "the calls sound lilke we're talking to darth vader" |
| 19:02.23 | [TK]D-Fender | That could do it on a bad design |
| 19:03.03 | devlap | What component should I mark in the jira ticket? |
| 19:03.14 | [TK]D-Fender | chan_sip |
| 19:05.01 | *** join/#asterisk gerritfromsa (~voipmagic@8ta-229-152-136.telkomadsl.co.za) |
| 19:05.26 | LemensTS | http://pastebin.com/DuJnMEXg here is cli output of the 2 polycom 321 phones that do not send BYE on hangup. This is with sip debug on for both phones |
| 19:05.37 | LemensTS | I waited 15seconds after hanging up to end the output |
| 19:05.39 | devlap | https://issues.asterisk.org/jira/browse/ASTERISK-23541 |
| 19:06.36 | *** join/#asterisk gentunian (~seba@unaffiliated/gentunian) |
| 19:09.43 | gentunian | I was wondering if someone could help me with call files... |
| 19:09.54 | [TK]D-Fender | gentunian: Ask away |
| 19:10.45 | gentunian | I can create call files for calling extensions, but seems I can't to dial an external number. |
| 19:10.59 | [TK]D-Fender | ~pb |
| 19:10.59 | 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. |
| 19:11.01 | [TK]D-Fender | ^^^^ |
| 19:11.07 | [TK]D-Fender | show us what you're doing |
| 19:11.14 | gentunian | ok |
| 19:12.29 | WIMPy | napnap: He, you finally found a straight cable? |
| 19:16.10 | gentunian | Here's the paste http://paste.lisp.org/display/141790 |
| 19:16.23 | gentunian | I have a typo in spoll |
| 19:16.43 | gentunian | in the paste not in the system. |
| 19:16.50 | gentunian | just in case... |
| 19:21.48 | napnap | WIMPy, hi, finally my cable was good...I don't know how finally it worked...but some test were "biased" because one of my two T0 is broken... |
| 19:22.48 | napnap | WIMPy, and now, after some successfull test I run in production mode but my t0 are down at this time :-( |
| 19:23.40 | napnap | WIMPy, but apparently this is due to software now....(I recompile dahdi at this time) |
| 19:24.15 | WIMPy | Fun with dahdi. The neverending story. |
| 19:24.54 | WIMPy | I still wonder if it has become worse or if I didn't test it propperly last time... |
| 19:24.56 | [TK]D-Fender | gentunian: show us the actual attempt with CLI & IAX2 debug enabled |
| 19:26.34 | WIMPy | And I was given the hint that I should test if the ptmp really works, which I admittadely didn't. |
| 19:27.42 | napnap | :-s , erf |
| 19:28.06 | napnap | I wish ptmp works |
| 19:29.16 | gentunian | [TK]D-Fender, I don't know if I understood that correctly, you mean what log says after trying? |
| 19:29.58 | [TK]D-Fender | gentunian: Yes |
| 19:31.31 | napnap | the moment of truth, compilation done (please I want to back home) |
| 19:32.03 | WIMPy | napnap: It will do enough ptmp for you. |
| 19:32.05 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 19:33.31 | gentunian | ok [TK]D-Fender http://paste.lisp.org/display/141791 |
| 19:33.36 | napnap | no it doesn't work :-\ |
| 19:35.02 | napnap | oh...perhaps |
| 19:35.19 | *** part/#asterisk nny (~Scott@174.107.218.2) |
| 19:36.06 | [TK]D-Fender | gentunian: the bottom one DID seemt o work. |
| 19:36.14 | [TK]D-Fender | It showed it answered and executed the playback |
| 19:36.35 | napnap | better ! yeah ! have some error like (Missing '/sys/bus/astribanks/drivers/xppdrv/sync') but my T0 works again ... I hope I don't need this file... |
| 19:36.43 | gentunian | yes, that it said inmediately, but the phone isnt ringing |
| 19:36.45 | gentunian | never rings |
| 19:36.58 | gentunian | maybe the trunk is not the adeccuate |
| 19:37.28 | napnap | (when I run dahdi service) |
| 19:37.55 | gentunian | I'm not the voip admin, I'm just trying to configure an alert with zabbix. So I'm gathering info by hand |
| 19:38.22 | [TK]D-Fender | gentunian: You din't enable IAX2 debug for your first call and the 2nd one looks good... |
| 19:38.29 | [TK]D-Fender | "iax2 set debug on" |
| 19:39.24 | gentunian | oh sry |
| 19:43.52 | gentunian | [TK]D-Fender, http://pastebin.com/4BKDddM1 |
| 19:44.07 | gentunian | I think the second looks good cause it calls to SIP/6001 |
| 19:44.15 | napnap | ok...I do nothing and my T0 are down ... |
| 19:44.34 | gentunian | if i try SIP/6051/<number> the last number isnt called. It only calls the extension 6051 |
| 19:44.42 | [TK]D-Fender | [Mar 26 16:41:13] VERBOSE[3800] chan_iax2.c: Timestamp: 00013ms SCall: 00635 DCall: 00000 [0.0.23.113:4569] |
| 19:44.46 | [TK]D-Fender | that IP doesn't look right... |
| 19:45.11 | gentunian | at all |
| 19:45.19 | gentunian | I didn't know that was an IP |
| 19:45.29 | [TK]D-Fender | Go look at your peer |
| 19:48.53 | gentunian | its strange, this is a working environment |
| 19:49.02 | gentunian | we can call outside |
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| 20:29.10 | *** join/#asterisk willwh (~willwh@unaffiliated/willskills) |
| 20:29.26 | willwh | hi guys, I'm trying to get sipml5 to play ball with asterisk 11.6-certified |
| 20:30.09 | willwh | think I am almost there, if I pastebin some logs, maybe someone's trying to do the same thing? :) |
| 20:31.25 | Qwell | ~ask |
| 20:31.25 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:32.13 | novaflash | [TK]D-Fender: i figure my issue out, i needed only to set nat=force_rport,comedia |
| 20:32.59 | willwh | actually, I got it working :D |
| 20:33.00 | willwh | yahoo |
| 20:37.19 | [TK]D-Fender | checkout time, BBIAB |
| 20:37.44 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
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| 21:14.39 | willwh | well, I'm closer: http://pastebin.com/eT3h9NKu |
| 21:15.51 | willwh | I don't know asterisk all that well yet, so if anyone has much experience with websockets / sipml5 / jsSip - I'd love to pick your brain |
| 21:16.13 | willwh | from what I have read in the docs for sipml5, and it's asterisk notes, this should work |
| 21:16.27 | willwh | I'm not trying to do video, or indeed webrtc, just plain sip with websockets |
| 21:16.30 | willwh | but using the sipml5 library |
| 21:16.57 | willwh | basically my call connects and I end up with: SetRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd. SIPml-api.js:1 |
| 21:17.27 | willwh | at a guess, this is maybe a lack of SSL? (that shouldn't matter surely as I'm using ws://) |
| 21:17.34 | willwh | not wss:// |
| 21:17.51 | willwh | Qwell: do you have any ideas? |
| 21:19.58 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 21:24.14 | *** join/#asterisk navaismo (~navaismo@201.124.126.200) |
| 21:25.53 | willwh | I see lots of chatter about this problem all over the place |
| 21:28.15 | newtonr | willwh, i believe there is already a couple of issues for that filed on the tracker. Are you using the latest SVN of 12 just to be sure? |
| 21:30.12 | willwh | no, I'm using 1.6-cert |
| 21:30.16 | willwh | I suppose I'm out of luck |
| 21:31.09 | [TK]D-Fender | 1.6.whatever = long-dead |
| 21:31.16 | willwh | roger |
| 21:31.26 | mjordan | 11.6 is what you're using |
| 21:31.36 | mjordan | try 11.8.1, as a number of ICE fixes went into recent releases |
| 21:31.57 | ChannelZ | You're as colllld as ice |
| 21:32.06 | willwh | willing to sacrifice |
| 21:32.07 | mjordan | I'm willing to sacrifice EVERYTHING |
| 21:32.12 | willwh | bwhahaha |
| 21:32.13 | willwh | :D |
| 21:32.19 | ChannelZ | :) |
| 21:32.58 | newtonr | willwh, if you can't get it working, throw your debug on ASTERISK-23425 or ASTERISK-23515, pretty sure those are duplicates of your issue |
| 21:33.22 | newtonr | willwh, that is, if your problem persists after 11.8.1 |
| 21:33.33 | willwh | yes, roger that, I'd guess it won't |
| 21:37.24 | willwh | thanks guys I really appreciate the guidance in new territory :) |
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| 22:03.12 | navaismo | is there a sangoma channel i cnt find it? |
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| 22:19.53 | navaismo | moy, did you now if wanpipe-7.0.10 compile on systems with kernel >= 3.7 |
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| 22:33.52 | willwh | hey guys :) |
| 22:34.02 | willwh | so I got a bit further, have 11.8.15 running now |
| 22:34.21 | willwh | same configs as 11.6, but the websocket handshake is failing now |
| 22:34.25 | willwh | 404ing |
| 22:34.32 | willwh | I see asterisk listening on 1060 |
| 22:37.08 | *** part/#asterisk thafreak (~thafreak@unaffiliated/thafreak) |
| 22:38.43 | navaismo | native connection or using a media gateway |
| 22:40.57 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 22:45.39 | willwh | sipml5 client |
| 22:45.50 | willwh | I connect over telnet to the port and get a 404 |
| 22:46.04 | willwh | just going to run over http sample config incase things changed from 11.6 -> 11.8 |
| 22:46.47 | *** part/#asterisk mjordan (~matt@nat/digium/x-hrczolxxgckgjcry) |
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| 22:52.56 | willwh | argh, my client died |
| 22:53.15 | willwh | so I only see, /httpstatic and /status as enabled URIs |
| 22:54.31 | willwh | odd, enabled=yes, and the bind address are set in http.conf |
| 22:54.34 | willwh | but I don't see /ws |
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| 22:59.47 | willwh | I wonder, ./configure --with-srtp=/usr/local/lib |
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| 23:10.16 | navaismo | i dont know what are you doing but if you provide configs and logs from js and asterisk we can help |
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| 23:14.23 | willwh | navaismo: oh, it's asterisk itself I'm having probs with |
| 23:14.27 | willwh | not the client |
| 23:14.42 | willwh | i.e. I have no res_http_websocket in the make menuselect list |
| 23:14.49 | willwh | I have res_srtp |
| 23:16.11 | willwh | <PROTECTED> |
| 23:16.36 | willwh | any ideas? |
| 23:16.52 | *** join/#asterisk lanning (~lanning@50.193.22.25) |
| 23:17.27 | navaismo | install the dependencies |
| 23:20.12 | willwh | navaismo: so I'm confused |
| 23:20.18 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 23:20.23 | willwh | I had it running np with 11.6 on the same system |
| 23:20.32 | willwh | i.e. at least websockets worked properly |
| 23:26.47 | navaismo | run the install_prereq script |
| 23:28.13 | *** join/#asterisk jmetro (ad0f27c3@gateway/web/freenode/ip.173.15.39.195) |
| 23:28.40 | jmetro | [TK]D-Fender: You there? |
| 23:29.46 | navaismo | willwh, then the configure script again |
| 23:30.14 | [TK]D-Fender | jmFor the next handful of minutes only... |
| 23:30.19 | [TK]D-Fender | jmetro: For the next handful of minutes only... |
| 23:30.48 | jmetro | [TK]D-Fender: nevermind then, i was going to ask a series of lengthy public IP / nat questions |
