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| 03:35.02 | linocisco | anybody who has tried asterisk external paging with grandstream phones? I tried according to Asterisk definitive guide 4th edition. ended with problem. |
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| 04:18.25 | tulga | multiple asterisk can use 1 E1 trunk? |
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| 04:25.39 | WIMPy | tulga: You can't connect multiple devices to an E1. |
| 04:26.56 | tulga | WIMPy: install 10 asterisks on 1 server with 1 E1, then share 30 concurrent calls? |
| 04:27.39 | WIMPy | What kind of sense does it make to install multiple Asterisks on one server? |
| 04:31.21 | tulga | WIMPy: I have 1 server with 1 E1, then I want service for install 3-4 companies |
| 04:31.32 | tulga | can I use 1 asterisk for multi tenant? |
| 04:31.41 | WIMPy | You don't need multiple Asterisks for that. |
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| 04:38.37 | tulga | WIMPy: great. I was checking thirdlane multi-tenant PBX and MIRTaPBX. is it what I want? |
| 04:43.14 | linocisco | anybody has done calling cards solution using asterisk? |
| 04:44.18 | WIMPy | tulga: I don't know what YOU want. |
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| 05:18.52 | [TK]D-Fender | linocisco: Tons of people |
| 05:20.43 | linocisco | [TK]D-Fender, If it is the call between only two countries and i am ok only with one originating countryand the other destination country has no easy negotiation |
| 05:21.26 | [TK]D-Fender | linocisco: That is a mangled question. |
| 05:22.33 | linocisco | [TK]D-Fender, thanks |
| 05:22.56 | [TK]D-Fender | Rephrase it. |
| 05:23.06 | [TK]D-Fender | I can't make sense of what you're trying to ask there. |
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| 08:11.05 | simpleTon` | hi i am building an asterisk based ip-pbx system with 10 incioming lines and 20 extensions.. how many ports trunk card should i buy |
| 08:18.28 | linocisco | simpleTon`, what 10 lines? landlines? |
| 08:38.41 | linocisco | hi all |
| 08:39.46 | linocisco | hi all, I tried to put this line exten => *723,1,Page(MulticastRTP/basic/239.0.0.1:1234) under "internal" dialplan and calling from one extension to another doesn't work anymore though that line was removed and asterisk and phones are restarted |
| 08:40.02 | linocisco | I tested with grandstream GXP1405 and zoiper |
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| 09:17.33 | gusto | hi |
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| 10:17.40 | Geek-Linux | Hi All, i want to know about asterisk video support and what would i need to implement the setup. ? |
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| 10:35.18 | gusto | so |
| 10:35.34 | boratynskikamil | Hello. I am preparing cluster based Call Center. How would you recommend. Divide 4 ISDN lines to 2 independent Asterisk PBX and each one is checking the state of devices on another one? |
| 10:35.56 | gusto | i just made asterisk 12 run with my custom set of modules |
| 10:36.38 | gusto | ha? |
| 10:38.08 | gusto | seems that all this pjsip replaced the modules which were asterisk-own ones and do not exist any more like callerid seems to be a part of pjsip now along with some SIP RFCs |
| 10:38.11 | gusto | so |
| 10:38.21 | gusto | then the bridges are extra |
| 10:38.24 | gusto | and now it works |
| 10:38.43 | gusto | but there are a lot more modules now |
| 10:39.35 | gusto | will chan_sip disappear completely now, or what? |
| 10:39.59 | gusto | i still have it in, and it does not seem to work without it, but it is somehow redundant now |
| 10:42.38 | file | chan_pjsip and chan_sip do not have feature parity, and can co-exist on the same system if they bind to different ports |
| 10:42.43 | file | so both will continue to exist |
| 10:43.08 | gusto | there is no chan_pjsip to me |
| 10:43.16 | gusto | i have only res_pjsip in there |
| 10:43.24 | file | it exists. |
| 10:43.37 | file | the PJSIP stuff is broken up into many many many modules |
| 10:43.55 | gusto | ah, i can load it, i just noticed it |
| 10:44.11 | gusto | however, i have pjsip not loaded now |
| 10:44.20 | gusto | you say that i can get rid of that pjsip res stuff? |
| 10:44.31 | file | if you aren't using PJSIP support you don't need to load any of the modules. |
| 10:44.37 | gusto | ah |
| 10:44.54 | gusto | and will my callerid work? because the old asterisk 11 callerid module is not there any more |
| 10:45.08 | file | what "old asterisk 11 callerid module"? |
| 10:45.12 | gusto | w8 |
| 10:45.41 | gusto | 'app_setcallerid.so' |
| 10:46.19 | file | that's been deprecated for, oh, 5 or more years now? |
| 10:46.24 | gusto | ok |
| 10:46.36 | file | it's still in the tree, though |
| 10:46.40 | gusto | so all this functions are now implemented into chan_sip, or where? |
| 10:46.45 | file | it's just not built by default |
| 10:46.47 | file | no |
| 10:46.48 | linocisco | hi all, for asterisk to asterisk over ADSL, what is the maximum bandwidth requirement for uplink speed /downlink speed for how many users |
| 10:46.49 | linocisco | ? |
| 10:47.08 | linocisco | hi all, for asterisk to asterisk over ADSL, what is the minimum bandwidth requirement for uplink speed /downlink speed for how many users |
| 10:47.14 | file | if you need that module then go into menuselect and enable it, it is not enabled by default because everyone has moved on to using func_callerid |
| 10:47.28 | gusto | you can count ... like 64kbps per user x 2 (up , down) |
| 10:47.54 | gusto | file: aha, so func_callerid is it, ok |
| 10:48.05 | file | func_callerid REPLACED it but it is not used in the same way |
| 10:48.30 | gusto | however, i have it in, so it must be right |
| 10:49.13 | gusto | i never checked that modules one by one, i moved from 1.8 with a list that i found on openwrt router package, because that one was good and fully functional |
| 10:50.07 | gusto | so basically the only difference now to the asterisk 12 is that they moved the bridge functions out |
| 10:50.25 | gusto | without that modules i was not able to make a call :D |
| 10:50.33 | file | we completely rewrote bridging. |
| 10:51.00 | file | and yes, you wouldn't be able to bridge calls without the modules to bridge them |
| 10:52.32 | gusto | ok, i kicked out that pjsip and it works |
| 10:52.57 | gusto | now i have only 55 modules |
| 10:53.01 | gusto | that is better readable |
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| 11:23.06 | Geek-Linux | Hi All, i want to know about asterisk video support and what would i need to implement the setup. ? |
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| 12:06.48 | gusto | btw |
| 12:07.43 | gusto | is there a possibility to tell him some kind of priority for the codecs so that he does not go and transcode something while there is a codec that could be used instead that would not need transcoding for making a connection |
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| 12:36.54 | davlefou | hi, somes thing is strange, i need to put canreinvite=no and nat=force_rport,comedia on my asterisk who is under server host. |
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| 12:57.36 | gusto | a very brutal way, but i proved it to be functional is to just remove every codec module that you do not want to be reencoded (into,out-to) and he is then taking the next best one |
| 12:58.11 | gusto | but i wonder if this causes some problems, for example because of retransmiting some messages through sip and so on, i ll have to debug that as well |
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| 13:10.43 | gusto | eh, as of before asterisk 12 he told me if he is locally bridging or remotely bridging, i even looked now at the bridge show command and not even there i can read if it is a local or remote bridge, the only way i see it is either over rtp set debug on or sip set debug on |
| 13:10.52 | gusto | there must be some other way to look into it |
| 13:11.46 | gusto | or local show channels, that may indicate something, i ll have to try a local bridge, maybe it will show up there |
| 13:13.42 | gusto | local show shows nothing |
| 13:14.09 | gusto | but there is a difference in the technology in native bridge being directmedia and simple bridge locally bridged one |
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| 13:31.46 | gusto | is the asterisk wiki down or something? |
| 13:32.12 | gusto | it does not seem to work, however |
| 13:41.53 | Qwell | gusto: It's working fine. |
| 13:42.15 | gusto | what? |
| 13:42.29 | gusto | not to me |
| 13:42.39 | gusto | is it somehow ipv6 enables w/o ipv6 delivering? |
| 13:45.30 | gusto | ok, i just installed w3m, because on graphical browsers you can not put an option which ipv protocol he should use and really, it loads under -4 but does not with -6 |
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| 13:45.48 | gusto | w3m says that he is stuck while opening socket using ipv6 |
| 13:46.03 | gusto | and somehow that crashed my w3m :-D |
| 13:46.30 | gusto | i can put it in a list of ipv4 only websites |
| 13:46.34 | gusto | on firefox |
| 13:47.30 | gusto | ok, i put it in ipv4 only domains, now it works, however, that is a quirk, one should fix that ipv6 connectivity |
| 13:47.50 | gusto | it looks like a firewall issue there |
| 13:48.41 | gusto | 2001:470:e0d4::ee answers to icmp, so that can not be the problem, maybe misconfiguration on a webserver? |
| 13:52.13 | gusto | a ssl problem maybe ... that he wants to redirect on ssl port and recieves no answer there, that's my unproven theory now, lol |
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| 13:54.38 | gusto | ... https://[2001:470:e0d4::ee]/ here, for those who are curious |
| 13:55.28 | gusto | port is open and responds, but fails somewhere at ssl |
| 13:55.40 | gusto | 443/tcp open https |
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| 15:31.14 | ipengineer | file: Are you around? |
| 15:31.23 | file | yes |
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| 15:33.14 | ipengineer | I was talking to one of my developers about this dialog-info issue and I am not sure we can easily do this without spending a LOT of time getting them up to speed with the asterisk core itself. That is where the breakdown will be.. We got to talking and we have Kamailio as a proxy between Asterisk and the UA. Could we just have kamailio look for a certain Notify event and add the dialog-info payload to that? If so that ma |
| 15:33.15 | ipengineer | a simple solution for us until this can get done correctly and implemented the correct way |
| 15:33.43 | file | I don't know Kamailio to that extent and can't answer that question. |
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| 15:34.06 | file | Probably not, though. |
| 15:37.36 | mjordan | Asterisk wouldn't know to raise the NOTIFY event |
| 15:38.18 | ipengineer | Can kamailio not subscribe for notify events? Maybe not dialog-info but some other format? |
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| 15:51.17 | ipengineer | file: I can post a bounty for this.. I know we discussed that initially. If someone picks it up that will save us both some work. I am more than willing to pay to get it done.. Do you have “rough” time estimates so I can come up with an offer? |
| 15:51.34 | file | that's a complicated question |
| 15:51.58 | ipengineer | I tend to be on a role with those here lately.. |
| 15:52.27 | file | you should ask mjordan! he's good at making up fictional numerical values |
| 15:52.35 | ipengineer | Haha ok.. |
| 15:53.52 | ipengineer | mjordan: Do you have any thoughts on how long it may take someone that is familiar with asterisk to build out this dialog-info functionality? |
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| 16:59.43 | jeffspeff | i'm getting the following sip debug output when a call is sent to a new sip trunk. http://pastebin.com/jLRctWc5 |
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| 17:01.06 | mjordan | ipengineer: familiar with the pjsip stack, not long. Not familiar with the pjsip stack, a bit longer. Particularly if they don't ask for help :-) |
| 17:01.35 | [TK]D-Fender | jeffspeff: that has nothing to do with a call. |
| 17:01.42 | mjordan | as for fictional numbers... let me see if I can't find a corollary |
| 17:02.39 | mjordan | file: does this only need a body generator? |
| 17:08.27 | mjordan | looks at file |
| 17:10.05 | mjordan | ipengineer: so my guess is (since file has gone 'dark') is that this is two modules, both approximately a few hundred lines of code. It's not super hard, as there are (a) examples of modules that do this already and (b) a review with a rejected approach available |
| 17:10.42 | mjordan | ipengineer: you'd need a module that subscribed for the dialog state in Asterisk (or otherwise monitored the state) and hooked itself onto res_pjsip_pubsub |
| 17:10.43 | file | my IRC disconnected for a sec :P |
| 17:10.57 | mjordan | ipengineer: and you'd need a module that was a body generator for the NOTIFY requests and file can correct me if I'm wrong |
| 17:10.59 | file | mjordan, no - it requires slight modification to allow body generators to store state |
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| 17:11.13 | mjordan | ah, so a bit of work in res_pjsip_pubsub then too |
| 17:11.31 | file | and a body generator |
| 17:11.35 | mjordan | file: can it use res_pjsip_extenstate? |
| 17:11.38 | file | and preferably not duplicating the res_pjsip_exten_state module |
| 17:11.42 | mjordan | right |
| 17:12.34 | mjordan | file: so you'd want to base the dialog state off of the device state subscription? |
| 17:13.48 | file | yes |
| 17:14.09 | file | dialog-info+xml is unlike other contents because it has an incrementing numerical value |
| 17:14.28 | file | (per subscription, must start at 1 if I remember right) |
| 17:14.37 | mjordan | ew |
| 17:14.53 | mjordan | but if it is per subscription, that's not terrible. |
| 17:15.12 | mjordan | Store a void * to some user defined ao2 object that they can pass via the body generator on subscription |
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| 17:15.51 | mjordan | although that could get a bit messy, since you'd have to have the body generators consume subscription state so that they could interpret the current state of the counter |
| 17:16.16 | mjordan | ipengineer: SO. Revision #2: it's one module (not a very large one), with some tweaks to some existing APIs in res_pjsip_exten_state and res_pjsip_pubsub |
| 17:17.09 | file | mjordan, the slightly annoying part is that subscriptions have datastore support for this already |
| 17:17.16 | file | (that's what I used last time) |
| 17:17.19 | file | but body generators are generic |
| 17:17.24 | mjordan | file: but we didn't expose them up through the API? |
| 17:17.54 | file | putnopvut kept body generators generic - they aren't sub specific |
| 17:21.25 | putnopvut | Correct. Body generators need to be usable for requests outside of subscriptions, such as PUBLISH and unsolicited MWI. |
| 17:23.17 | davlefou | hi, i have that error : chan_sip.c:4343 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data |
| 17:23.25 | davlefou | i try to use sql |
| 17:23.26 | file | walks home |
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| 17:37.15 | file | awww I missed nothing |
| 17:44.35 | ipengineer | file: mjordan Sorry.. I just got back from a meeting. Ok I think that clears it up a little bit |
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| 17:58.37 | tm1000 | can feature codes be detected in app_meetme or app_confbridge |
| 17:58.46 | davlefou | it seems an problem with sip.conf put un static |
| 17:58.48 | tm1000 | such as *<dtmf> |
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| 18:17.14 | file | tm1000, no |
| 18:17.21 | file | tm1000, not unless you front end your call with a Local channel |
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| 18:37.36 | moy | willwh: did you get wss working? |
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| 18:45.00 | willwh | moy: ! zomg |
| 18:45.02 | willwh | no :( |
| 18:45.07 | willwh | I didn't have time to file a bug |
| 18:45.23 | WIMPy | You get used to that. |
| 18:45.52 | willwh | it's interesting, in a js console; conn = new WebSocket('http://blabla:10060/ws','sip') |
| 18:45.59 | willwh | conn.readyState = 1 |
| 18:46.17 | willwh | but if I go, https://blabla:10060/ws - no dice |
| 18:46.28 | willwh | my cert is ok, I enabled the AMI http interface with tls in manager.conf |
| 18:46.33 | willwh | and it is a kosher cert |
| 18:47.01 | willwh | and oops, the WebSockets are like (ws://server:10060/ws) and (wss://) respectively |
| 18:47.13 | willwh | when trying wss I just get "3" which is closed |
| 18:47.49 | willwh | moy: what timezone are you on? I'm at work atm, and I don't have much time to chat |
| 18:48.04 | willwh | I'm on Pacific Standard Time |
| 18:48.28 | willwh | I'm running the very latest 12.2.0-RC-2 from svn |
| 18:48.32 | willwh | which I am pretty sure has your changes |
| 18:48.43 | willwh | I didn't bring my laptop to work or I could fire up my dev env |
| 18:49.00 | willwh | moy: ^ :) |
| 18:52.38 | willwh | moy: when issuing 'http show status' - should I expect to see anything about Secure ws? |
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| 19:42.54 | willwh | moy: give me a shout when you can, I'll idle in here :) (woot irssi on a vps) |
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| 19:52.08 | davlefou | i find the problem, ip error |
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| 20:43.31 | lordvadr | Is there a version of "core show channels" that lists the full channel names? Longer channel names gets truncated by the output formatting. From a different perspective, I'm writing a script and I need to find all the channels that are up and have called some number. "asterisk -rx 'core show channels' | grep <number>" kinda gets there, but often times part of the channel names are truncated. |
| 20:44.54 | WIMPy | core show channels <tab> |
| 20:45.20 | lordvadr | WIMPy: Can't script that. |
| 20:45.54 | WIMPy | Have you tried doing it yopurself? |
| 20:46.30 | lordvadr | doing what myself? Actually, "core show channels concise" appears to be exactly what I want. I've used the verbose version but didn't know that "concise" existed. |
| 20:46.52 | WIMPy | See, that's why you should press tab. |
| 20:46.59 | willwh | yay tab completion >.> |
| 20:47.29 | eirirs | tab symbol on my tab are gone. |
| 20:47.33 | eirirs | tab abuse <3 |
| 20:47.35 | lordvadr | I've hit tab after that several times, but when a server responds with a couple hundred channels, the "concise" part always got lost. |
| 20:48.06 | eirirs | core show channels c<tab> works too |
| 20:48.08 | WIMPy | No. |
| 20:48.23 | WIMPy | That would have been core show channel <tab> |
| 20:48.41 | lordvadr | WIMPy: Yeah, that's exactly what happened. Thanks for that. |
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| 20:59.20 | moy | willwh: you should at least see HTTPS server enabled |
| 21:00.14 | moy | willwh: you should read the troubleshooting hints on this presentation: http://goo.gl/iJyaXZ |
| 21:00.48 | moy | starting from slide 16 particularly |
| 21:02.38 | moy | willwh: regarding tz, I am on EDT |
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| 21:10.48 | shuggans | Hi all - Wondering if anyone can help me out with getting nortel 1110 phones working with asterisk/freepbx |
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| 21:23.22 | newtonr | shuggans, depending on the issue, yes, someone can likely help. If you are using FreePBX. You may get a better answer in the #freepbx channel |
| 21:23.30 | newtonr | ~ask |
| 21:23.30 | 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. |
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| 22:19.06 | willwh | moy: cool, well, I will crank on this tonight, you'll probably be sleeping :) |
| 22:19.27 | willwh | moy I should say, I have gone through that :) |
| 22:19.42 | willwh | If I don't get it, should I bug report, or just mailing list? |
| 22:20.35 | willwh | moy: thanks so much for reaching out to me though :) |
| 22:20.47 | willwh | I'll write up my configs / deps installed etc |
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| 22:22.10 | willwh | ah, moy I see my prob |
| 22:22.38 | willwh | I guess it helps to read, re-read, then come back another day and read it again, haha, I don't have tlsenable/tlsbindaddr set in the http.conf |
| 22:22.42 | willwh | shoots himself |
| 22:22.56 | willwh | I'd imagine this will work when I get home, but I'll ping you and let you know |
| 22:23.06 | willwh | thank you for all your work, just awesome! :) |
| 22:23.54 | willwh | moy++ |
| 22:26.13 | willwh | this needs an update: https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support :) |
| 22:26.16 | willwh | I'll try and do that too |
| 22:27.33 | navaismo | that is ok for WS only |
| 22:27.47 | willwh | navaismo: yeah, but it should have a section for wss |
| 22:27.58 | willwh | 'cause I think all that I need is tls details in http.conf, and 12.2.0-rc2 will work |
| 22:28.16 | navaismo | and just to be curious did you know soon chrome will not work with asterisk because the recent change with chrome |
| 22:28.22 | willwh | I did not\ |
| 22:28.28 | willwh | which is what? |
| 22:28.39 | willwh | navaismo: got a link? |
| 22:28.51 | navaismo | 1 sec |
| 22:28.58 | willwh | appreciate it thx |
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| 22:30.12 | navaismo | https://code.google.com/p/webrtc/issues/detail?id=2774 |
| 22:31.35 | navaismo | so if you are "investing" time creating an app you need to consider or a MediaGateway or wait for an official RFC for Webrtc this days the hype is mess |
| 22:32.38 | navaismo | the Guys from Jssip wants to rule the world with their RFC, in the meantime others are stuck in a battle between SDP or not SDP and finally others with VP8 and open-H264 |
| 22:32.57 | navaismo | so basically webrtc turns in a Lego kit for recreation |
| 22:33.28 | navaismo | ok let me rephrase that... WebRTC apis |
| 22:35.17 | navaismo | https://issues.asterisk.org/jira/browse/ASTERISK-23425?filter=-3 |
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| 22:57.25 | joobie | hey guys, what variable can i use to identify the dialling (src) extension ? |
| 23:01.38 | [TK]D-Fender | joobie: "core show function CDR" <- |
| 23:02.17 | joobie | ta [TK]D-Fender |
| 23:08.15 | marceloamorim | guys, I'm installing the asterisk on the debian, and I'm following the wiki.asterisk.og, I'm on the install_prereq unpackeg and I don't have subversion installed, so I installed and do the command "./install_prereq isntall-unpackeged", but now it takes so long |
| 23:08.19 | marceloamorim | this is normal? |
| 23:09.09 | marceloamorim | I was checking the the install_prereq and this script is so simple, just svn the nbs-trunk, make & make isntall |
| 23:15.10 | marceloamorim | never mind, I killed the process and run again, now it did faster |
| 23:15.12 | sectech | Not interested in starting a flamewar, but is there really that much different between Asterisk and Freeswitch? |
| 23:15.35 | sectech | I kinda need to dedicate my efforts towards one... not both |
| 23:15.51 | sectech | I know asterisk fairly well, not much about freeswitch though |
| 23:17.32 | sectech | If you try and google the differences, you just find the pissing matches that have gone on in the past... Not very much useful info |
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| 23:24.07 | file | falls into existence |
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