| 00:03.34 | *** join/#asterisk SuperBawlz (~zbriggs@rrcs-71-43-76-226.se.biz.rr.com) |
| 00:03.41 | SuperBawlz | anyone home? |
| 00:14.17 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 00:26.25 | SuperBawlz | Bueler?.....anyone......anyone..... |
| 00:26.35 | cusco | hi |
| 00:26.42 | cusco | i'm at home |
| 00:29.58 | SuperBawlz | LOL |
| 00:30.05 | SuperBawlz | So I have a funny one. |
| 00:30.10 | cusco | shoot |
| 00:30.32 | SuperBawlz | I have three systems with sip trunks using the same provider |
| 00:30.44 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.122) |
| 00:30.46 | cusco | ok.. |
| 00:31.08 | SuperBawlz | two are 1.8.9.0 and 1.8.17.0 |
| 00:31.17 | SuperBawlz | one is 1.8.21.0 |
| 00:32.06 | SuperBawlz | the two are able to use trunk settings with this command present "allow=none&ulaw" |
| 00:32.58 | SuperBawlz | but if the command is used on 1.8.21.0 outbound calls drop after 8 seconds. Only the audio |
| 00:33.36 | SuperBawlz | that's pretty jacked up eh? |
| 00:35.29 | cusco | but the coddec is accepted? |
| 00:36.03 | cusco | I always used a allow line per codec |
| 00:36.06 | cusco | disallow=all |
| 00:36.14 | cusco | allow=alaw |
| 00:36.18 | cusco | allow=ulaw |
| 00:36.25 | cusco | but that should make no difference |
| 00:48.24 | igcewieling | pastebin the cli output of a failed call with sip debug enabled for the peer and hope for the best. |
| 00:48.40 | igcewieling | the allow=none is invalid anyway |
| 00:52.25 | SuperBawlz | cusco, using |
| 00:52.32 | SuperBawlz | disallow=all |
| 00:52.39 | SuperBawlz | allow=ulaw |
| 00:52.40 | SuperBawlz | works |
| 00:52.47 | cusco | ok then |
| 00:52.49 | cusco | sorted! |
| 00:55.11 | SuperBawlz | yeah. we figured it out but its weird. |
| 00:55.53 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.122) |
| 00:56.21 | *** join/#asterisk ircNewbz (~chatzilla@unaffiliated/ircnewbz) |
| 00:57.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.122) |
| 00:59.22 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.122) |
| 01:02.47 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.122) |
| 01:06.48 | sareebro | anyone know how to display page numbers with fax for asterisk? |
| 01:07.15 | cusco | display them...where? |
| 01:07.20 | sareebro | in the header |
| 01:07.38 | cusco | I think there's a variable for that |
| 01:09.33 | cusco | FAXPAGES |
| 01:09.35 | cusco | :) |
| 01:09.50 | cusco | NoOp(Fax ${NUMBER} ${CALLERID(num)} with ${FAXPAGES} pages, ${FAXBITRATE} bitrate, status: ${FAXSTATUS} error: ${FAXERROR} and resolution: ${FAXRESOLUTION}); |
| 01:13.05 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 01:13.51 | sareebro | cusco: I looked it up in the admin manual, but FAXPAGES is the number of fax pages transmitted, from how I interpreted it... |
| 01:13.59 | sareebro | doesn't sound like something I could use for the header itself |
| 01:14.02 | cusco | ow... |
| 01:14.20 | cusco | I don't understand |
| 01:14.30 | cusco | what is it that you want |
| 01:15.33 | sareebro | I want the fax to show page numbers in the header. |
| 01:17.43 | igcewieling | sareebro: inbound or outbound faxing? |
| 01:18.06 | sareebro | igcewieling: outbound |
| 01:18.33 | igcewieling | I missed your actual question. What is the question? |
| 01:19.02 | sareebro | igcewieling: is there a way to display page numbers in the header when faxing with Fax For Asterisk? |
| 01:20.02 | igcewieling | I believe you can set the header. You'll have to figure out the number of pages using some other method like using tiffinfo or similar linux command |
| 01:20.08 | cusco | display in the tiif file? |
| 01:20.12 | cusco | like, in the image? |
| 01:20.19 | cusco | I don't thin asterisk has a way of adding data |
| 01:20.26 | cusco | but you can post process it with some file |
| 01:20.29 | cusco | some tool |
| 01:20.34 | cusco | like imagemagick or something |
| 01:21.27 | igcewieling | FAXOPT(headerinfo) or FAXOPT(localstationid) |
| 01:21.30 | *** join/#asterisk j-fish (~unrea@unaffiliated/j-fish) |
| 01:21.42 | igcewieling | likely headerinfo |
| 01:22.05 | sareebro | headerinfo is actually a variable that I am setting |
| 01:22.28 | sareebro | I was just wondering if there was an option I could turn on/off to display |
| 01:22.50 | sareebro | spandsp seems to automatically print the date/time as well as the page number in addition to the headerinfo that i set |
| 01:23.35 | j-fish | My question is not related to asterisk (although i use it) but maybe someone here can actually help,i send and recieve around 2000 text messages a day , and it's not working good with regular cell companies,are there any alternatives? |
| 01:23.41 | sareebro | I was just curious. I suppose I will try using another utility to modify it. Thanks cusco and igcewieling. |
| 01:24.37 | igcewieling | j-fish: there are commercial companies which provide bulk SMS services. j-fish what country are you in |
| 01:24.38 | igcewieling | ? |
| 01:24.54 | j-fish | igcewieling: usa |
| 01:25.36 | igcewieling | j-fish: a commercial service is your only option. ALL Asterisk SMS related things only work with non-usa providers which provide SMS services over analog lines. |
| 01:26.05 | j-fish | i am running a call center and dispatch a lot of work to subconturcts through texts and they reply all day long |
| 01:26.30 | j-fish | igcewieling: i've been using vitielity sms for testing but it's not reliable |
| 01:27.30 | igcewieling | I don't use viteliry for SMS, but I've had an account with them since....2005 or earlier. |
| 01:27.39 | igcewieling | they seem to be pretty reliable for voice and fax |
| 01:27.49 | j-fish | android based phone with some app like allmighty text is very cheap and unlimited texts,but it's not reliable either |
| 01:27.56 | igcewieling | there are other, similar SMS providers out there. |
| 01:28.17 | j-fish | yeah they're excellent for voice |
| 01:30.00 | igcewieling | I doubt you'll ever get VERY reliable SMS service from anyone, but try a few different providers |
| 01:30.48 | j-fish | igcewieling: SMS is like fax,both should be gone from our life |
| 01:31.14 | igcewieling | j-fish: and neither will be for decades |
| 01:31.31 | j-fish | yep lol |
| 01:32.09 | j-fish | well thanks i will keep looking:) |
| 01:33.10 | cusco | well |
| 01:33.11 | igcewieling | You can be assured that "cheap" will not be reliable when it comes to SMS (even more so than with other services) |
| 01:33.18 | cusco | there is also the chan_dongle |
| 01:33.40 | cusco | where you connect a huawei usb dongle and send sms from the SIM in it |
| 01:34.00 | igcewieling | cusco: same issues, the carrier won't care about an end user non-commercial account |
| 01:34.31 | j-fish | igcewieling: can i get a commerical account with it? |
| 01:34.41 | cusco | igcewieling: I fail to see how that is a issue... |
| 01:35.29 | igcewieling | j-fish: with 40,000 texts/month any carrier will laugh at you if you ask about a commercial account. |
| 01:35.46 | igcewieling | cusco: he wants reliable. |
| 01:35.50 | j-fish | not to mention that i need to run it on at least 3 computers at the time (1 sms number to sync all outgoing/received messages) |
| 01:36.30 | cusco | igcewieling: I relly on my carrier when I send sms from my phone |
| 01:36.32 | cusco | but then again |
| 01:36.38 | cusco | only now Inoticed the volume he meant |
| 01:36.41 | cusco | 2000 a day |
| 01:37.29 | igcewieling | j-fish: what is your failure rate on Vitelity SMS? |
| 01:37.34 | j-fish | well sprint are reliable for us,its not as much of the carrier as the program we're using i guess |
| 01:38.06 | j-fish | igcewieling: i didnt check the rate,just saw a lot of texts not showing up and other issues that they have not fixed yet |
| 01:38.22 | cusco | we use a service from the biggest telco in our country, costs 0.05EUR per sms |
| 01:38.26 | cusco | soap |
| 01:38.46 | igcewieling | j-fish: Do all of the people receiving the SMSs have smartphones? Coding an app for android and Apple might not be a bad thing to consider, then you can avoid the SMS issue entirely. |
| 01:39.13 | cusco | it supports so much stuff, delivery reports check, sms with extra cost etc |
| 01:39.40 | j-fish | igcewieling: no we have so many subcontractors i can not ask them all to have smartphones |
| 01:40.35 | igcewieling | cusco: Providers Europe is totally different thing from SMS is the USA |
| 01:40.56 | cusco | ok igcewieling |
| 01:41.27 | igcewieling | cusco: carriers in the USA to not provide outside access to their network using stuff like SOAP. |
| 01:41.47 | igcewieling | They also don't allow access to their SMSC |
| 01:41.55 | cusco | ew |
| 01:42.02 | cusco | eww! |
| 01:42.04 | cusco | ok ok |
| 01:42.11 | cusco | I don't wan't to know more.P |
| 01:42.12 | cusco | :P |
| 01:42.27 | igcewieling | cusco: in many ways cell companies in the USA are very behind. |
| 01:42.35 | igcewieling | same with ISPs in the USA |
| 01:42.58 | cusco | ok |
| 01:43.13 | igcewieling | You can't, for example, go town to your local corner store and buy a SIM card in the USA |
| 01:43.32 | igcewieling | in fact the 2 largest carriers don't even use SIM cards. |
| 01:43.52 | igcewieling | sorry, largest and 3rd largest carriers |
| 01:44.00 | cusco | what do they use? |
| 01:46.50 | igcewieling | CDMA, which does not use SIM cards and the phone cannot generally be moved between carriers. |
| 01:47.23 | igcewieling | This is changing in the future as carriers move their voice services over to LTE, but it will be a few years before that starts to happen. |
| 01:48.04 | cusco | here we have lte, based on gsm |
| 01:48.12 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 01:48.21 | igcewieling | AT&T and T-Mobile use GSM and SIM cards. Verizon and Sprint use CDMA which does not even support the concept of SIM cards. |
| 01:48.36 | cusco | ok.. |
| 01:48.40 | *** join/#asterisk vlad_starkov (~vlad_star@87.245.131.34) |
| 01:50.02 | igcewieling | Even carriers with LTE still use GSM (or CDMA) for voice, though voice will be moved to LTE in the future. |
| 01:50.24 | cusco | ow.. ok |
| 01:50.36 | igcewieling | this is the case globally, not just in the USA |
| 01:50.40 | cusco | say |
| 01:50.50 | cusco | lte uses ipv6 & ipv4 dualstack |
| 01:50.52 | cusco | right? |
| 01:50.59 | cusco | in the standard |
| 01:51.21 | igcewieling | I'm not sure, but I don't think so. |
| 01:54.14 | *** part/#asterisk ircNewbz (~chatzilla@unaffiliated/ircnewbz) |
| 01:57.34 | *** join/#asterisk vlad_sta_ (~vlad_star@87.245.131.34) |
| 01:58.13 | *** join/#asterisk vlad_sta_ (~vlad_star@87.245.131.34) |
| 02:06.00 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 02:08.02 | *** part/#asterisk j-fish (~unrea@unaffiliated/j-fish) |
| 02:25.30 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 02:39.42 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 02:50.02 | *** join/#asterisk inaeth (~inaeth@unaffiliated/inaeth) |
| 02:55.25 | *** join/#asterisk FonJocke_ (~FonJockey@m822c36d0.tmodns.net) |
| 02:55.32 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 02:56.49 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.130) |
| 03:04.12 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.124.101) |
| 03:41.47 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 03:53.45 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.72) |
| 04:02.24 | *** join/#asterisk bkruse (~Adium@107-220-153-105.lightspeed.jcvlfl.sbcglobal.net) |
| 04:08.43 | *** join/#asterisk FonJockey (~FonJockey@173-133-156-123.pools.spcsdns.net) |
| 04:11.29 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 04:27.48 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
| 04:44.11 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.170.25) |
| 04:45.06 | *** join/#asterisk FonJocke_ (~FonJockey@mbd2c36d0.tmodns.net) |
| 05:06.39 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
| 05:07.17 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 05:13.39 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
| 05:18.43 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 05:24.57 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 05:25.05 | linocisco | hi all |
| 05:25.25 | linocisco | allow=alaw,ulaw,gsm or allow=alaw&ulaw&gsm? |
| 05:25.37 | linocisco | which one is correct? |
| 05:28.10 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
| 05:28.19 | ectospasm | linocisco: the commas |
| 05:28.27 | linocisco | ectospasm, thanks |
| 05:28.41 | ectospasm | or, list each codec separately with different allow= lines |
| 05:28.44 | linocisco | ectospasm, gsm or gsmlaw? |
| 05:28.55 | ectospasm | gsm |
| 05:28.57 | ectospasm | ulaw |
| 05:28.59 | ectospasm | alaw |
| 05:29.00 | linocisco | ectospasm, thanks |
| 05:32.51 | *** join/#asterisk cmendes0101 (~cmendes01@96.247.7.238) |
| 05:36.44 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 05:54.17 | igcewieling | generally you don't want to allow both ulaw and alaw |
| 05:54.46 | igcewieling | linocisco: freepbx translates the & to individual allow lines |
| 05:55.13 | linocisco | igcewieling, i m refering http://www.youtube.com/watch?v=xZU339bzkZw |
| 05:55.24 | linocisco | igcewieling, but can't get it to work. |
| 06:06.10 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 06:16.40 | *** join/#asterisk _mnathani_ (~mnathani@198-84-231-11.cpe.teksavvy.com) |
| 06:23.54 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 06:40.07 | *** join/#asterisk iprouteth0 (~james@unaffiliated/iprouteth0) |
| 07:03.57 | *** join/#asterisk evilman_home (kvirc@128-73-177-130.broadband.corbina.ru) |
| 07:23.42 | *** join/#asterisk mihamina (~mihamina@static-110-9.blueline.mg) |
| 07:23.45 | *** part/#asterisk mihamina (~mihamina@static-110-9.blueline.mg) |
| 07:31.21 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 07:31.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 07:57.21 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 08:04.29 | *** join/#asterisk MrTAP (~Trev@d108-180-49-225.bchsia.telus.net) |
| 08:08.33 | *** join/#asterisk mathi (Matt@92.249.121.92) |
| 08:12.04 | *** join/#asterisk roentgen (~arthur@openvpn/community/support/roentgen) |
| 08:18.26 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 08:18.38 | *** join/#asterisk f843d0 (~f843d0@host68-125-dynamic.0-87-r.retail.telecomitalia.it) |
| 08:54.05 | *** join/#asterisk elmargol (~elmargol@host58-105-dynamic.8-87-r.retail.telecomitalia.it) |
| 09:29.31 | *** join/#asterisk izbushka (~izbushka@193.23.225.11) |
| 09:35.02 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 09:36.29 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 09:53.41 | *** join/#asterisk ghost75 (~trechber@dslb-088-066-167-054.pools.arcor-ip.net) |
| 10:38.46 | *** join/#asterisk jetlag (~jetlag@pool-71-168-242-55.cmdnnj.east.verizon.net) |
| 11:00.38 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 11:28.13 | *** join/#asterisk bkruse (~Adium@107-220-153-105.lightspeed.jcvlfl.sbcglobal.net) |
| 11:33.05 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.84) |
| 11:39.11 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 11:42.07 | *** join/#asterisk bkruse (~Adium@107-220-153-105.lightspeed.jcvlfl.sbcglobal.net) |
| 12:12.16 | *** join/#asterisk iprouteth0 (~james@unaffiliated/iprouteth0) |
| 12:33.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.144) |
| 12:37.25 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 13:14.14 | *** join/#asterisk mihamina (~mihamina@static-110-9.blueline.mg) |
| 13:20.35 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 13:24.08 | *** join/#asterisk bkruse (~Adium@107-220-153-105.lightspeed.jcvlfl.sbcglobal.net) |
| 13:27.57 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.131) |
| 13:53.00 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.193) |
| 13:53.12 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:53.12 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:55.01 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 13:55.23 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 14:13.19 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@ip4dab53a2.direct-adsl.nl) |
| 14:13.58 | *** join/#asterisk tzafrir (~tzafrir@bzq-84-109-18-138.red.bezeqint.net) |
| 14:15.56 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 14:19.27 | *** join/#asterisk andy09usa (~Andrey@unaffiliated/andy09usa) |
| 14:28.30 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:39.27 | *** join/#asterisk serafie (~erin@205.201.235.1) |
| 14:48.25 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:53.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.193) |
| 15:07.01 | *** join/#asterisk tzafrir (~tzafrir@bzq-84-109-18-138.red.bezeqint.net) |
| 15:08.07 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.193) |
| 15:09.08 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 15:10.24 | *** join/#asterisk gusto (~gusto@ppp-212-114-228-13.dynamic.mnet-online.de) |
| 15:10.56 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.91) |
| 15:12.36 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.91) |
| 15:13.58 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:14.37 | *** join/#asterisk bkruse (~Adium@107-220-153-105.lightspeed.jcvlfl.sbcglobal.net) |
| 16:14.56 | *** part/#asterisk subthalamus (~chatzilla@bou57-1-78-247-40-105.fbx.proxad.net) |
| 16:23.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.169) |
| 16:30.13 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.169) |
| 16:36.49 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 16:37.39 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 16:41.05 | *** join/#asterisk linocisco (~linocisco@203.81.72.81) |
| 16:41.08 | linocisco | hi all |
| 16:41.57 | linocisco | i heard asterisk can't let users or extensions press 91 and hear dial tone or busy tone to check if line is free or not |
| 16:42.59 | linocisco | so alternatively, can asterisk make cisco ip phone show as FXO line indicator to see whether FXO line is occupied or not before making a call. |
| 16:43.24 | linocisco | otherwise, line busy may be in two forms, our source FXO line is busy or destination number is busy |
| 16:44.13 | igcewieling | linocisco: in SIP digits are not sent to asterisk until the user completes dialing on the phone. If asterisk doesn't get the digits there is not much Asterisk can do. |
| 16:45.33 | igcewieling | this is fundamental concept about how SIP works and you MUST understand it. |
| 16:45.57 | linocisco | igcewieling, ok. so if we want to call outside, is it ok to press to dial 91 and asterisk can see line is busy or not? if FXO line 1 is busy, will it automatically procceed to next line 2 ? |
| 16:46.23 | igcewieling | linocisco: yes it is possible in theory to monitor the state of a device/port but you are not even close to understanding Asterisk enough to do that and you are using FreePBX which makes it even more complicated. |
| 16:46.50 | linocisco | igcewieling, yes. that is why I formatted freepbx |
| 16:46.51 | igcewieling | linocisco: if you configure your phone to send digits after dialing 91 then you can handle it in the dialplan |
| 16:46.52 | linocisco | HDD |
| 16:47.20 | igcewieling | if the port is busy then Dial will set DIALSTATUS to CHANUNAVAIL, not BUSY |
| 16:47.58 | linocisco | igcewieling, will it move to next available one automatically ? |
| 16:48.27 | igcewieling | linocisco: it will move to the next priority yes. |
| 16:48.41 | igcewieling | as for moving to the next available port, that is a function of your SIP gateway, not Asterisk |
| 16:49.25 | linocisco | igcewieling, all explanations on most books are using DAHDI and digium cards. mine is grandstream FXO gateway device which accept 4 PSTN lines in maximum. i found no explanation on that |
| 16:49.58 | igcewieling | linocisco: that is because as far as asterisk is concerned you don't have FXO ports. You simply have a sip peer. |
| 16:50.25 | igcewieling | once the call gets to the sip peer aka device it is up to the device to handle selecting ports and hunting, etc, not asterisk |
| 16:51.38 | linocisco | igcewieling, what I was facing is that call can be heard from one side if we dialed from users behind asterisk. for incoming calls from PSTN into asterisk , call were fine |
| 16:51.40 | igcewieling | Asterisk doesn't know if your sip device is a phone, a gateway, or a trunk to a provider -- they are all just sip peers. |
| 16:54.47 | linocisco | igcewieling, what I was facing is that call can be heard from one side only if we dialed from users behind asterisk. for incoming calls from PSTN into asterisk , call were fine |
| 16:55.26 | igcewieling | one-way audio issues are typically nat issues |
| 16:55.52 | igcewieling | they could also be codec issues, but that is uncommon |
| 16:57.22 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 17:08.53 | linocisco | igcewieling, hi |
| 17:09.44 | linocisco | igcewieling, we are not calling over inside out of NAT. PBX is setup for PSTN inside out call |
| 17:14.42 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 17:28.45 | igcewieling | linocisco: pastebin the cli output with sip debug enabled for the peer and hope someone can help. |
| 17:29.22 | linocisco | igcewieling, thanks |
| 17:51.11 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@ip4dab53a2.direct-adsl.nl) |
| 18:15.31 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 18:26.42 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 18:35.19 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@ip4dab53a2.direct-adsl.nl) |
| 18:37.15 | *** join/#asterisk FonJockey (~FonJockey@108.107.116.201) |
| 18:40.58 | *** join/#asterisk sevin (~Thaddeus@c-50-132-126-144.hsd1.wa.comcast.net) |
| 18:45.12 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 18:48.13 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-231-146.ph.ph.cox.net) |
| 18:54.11 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 19:00.08 | *** join/#asterisk bchia (~Adium@75.76.51.204) |
| 19:02.15 | bchia | Anyone familiar with starpy? I'm getting "WARNING:FastAGI:Line received without pending deferred: 'HANGUP'" when trying to call agi.hangup() and the channel is still up. How do I hangup a channel? |
| 19:03.10 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.61) |
| 19:43.22 | *** join/#asterisk TimeRider (~steve@90.213.18.65) |
| 19:44.15 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:44.15 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:51.44 | *** join/#asterisk navaismo (~navaismo@189.241.67.184) |
| 19:54.30 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.61) |
| 19:58.46 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@ip4dab53a2.direct-adsl.nl) |
| 20:00.12 | *** part/#asterisk navaismo (~navaismo@189.241.67.184) |
| 20:00.58 | *** join/#asterisk iprouteth0 (~james@unaffiliated/iprouteth0) |
| 20:01.52 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 20:01.52 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 20:05.33 | bchia | For anyone who might find this later (searching logs, etc…) the solution I found was to use agi.finish() instead of agi.hangup() - this closed that one connection (and hungup the channel) while leaving the reactor running to receive more calls. |
| 20:11.41 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 20:35.25 | *** join/#asterisk voxter_ (~voxter@d23-16-70-23.bchsia.telus.net) |
| 20:45.00 | *** part/#asterisk bchia (~Adium@75.76.51.204) |
| 20:56.42 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 20:59.04 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-231-146.ph.ph.cox.net) |
| 21:19.56 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 21:24.05 | *** join/#asterisk voxter_ (~voxter@d23-16-70-23.bchsia.telus.net) |
| 21:24.07 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.109) |
| 21:51.47 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 21:52.33 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 21:56.53 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 22:09.26 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
| 22:10.10 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 22:20.30 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@ip4dab53a2.direct-adsl.nl) |
| 22:29.21 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 22:46.04 | *** join/#asterisk alhemicar (~alhemicar@178.77.1.189) |
| 23:24.43 | *** join/#asterisk MarKsaitis (~MarKsaiti@2.217.210.210) |
| 23:25.11 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.230) |
| 23:31.46 | *** part/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 23:38.28 | Kobaz | anyone know if polycom phones support third party call control directly |
| 23:39.26 | igcewieling | Kobaz: define "thirs party call control" |
| 23:40.08 | Kobaz | tell the phone to do things, from outside the phone, and outside of just plain sip signalling |
| 23:40.11 | Kobaz | oooOoo |
| 23:40.41 | Kobaz | <PROTECTED> |
| 23:40.51 | igcewieling | yes. |
| 23:40.59 | Kobaz | i guess it used to be a license-only feature |
| 23:41.15 | Kobaz | now the question is, where's the docs on how to use it |
| 23:41.21 | igcewieling | there are multiple methods to control the phone remotely. |
| 23:42.03 | Kobaz | now i know what it's called, uaCSTA |
| 23:42.58 | igcewieling | that is just one method |
| 23:43.03 | igcewieling | look for "push" in http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/Developers_Guide_UCS_4_0_1.pdf |
| 23:43.51 | igcewieling | there is also broadsoft and microsoft linc integration, I assume those support some form of call control as well |
| 23:44.11 | Kobaz | yeah linc and etc wont help me |
| 23:44.18 | Kobaz | i want call control for unit testing |
| 23:44.55 | igcewieling | so something like "A push request is defined as an XML formatted request that you send to a phone to tell it to process the XML content. The phone may render the data, fetch a URL, or perform an action." |
| 23:45.14 | Kobaz | yeah i know about polycom push |
| 23:45.23 | Kobaz | didn't know you can use it to do cal lcontrol |
| 23:45.29 | Kobaz | i've used it for just messaging and stuff |
| 23:45.55 | igcewieling | I'm pretty sure you can tell it to push buttons for you, but you'd have to check to be sure. |
| 23:46.03 | Kobaz | wow this hotel wireless is really getting crappy, i might be better off on tether |
| 23:46.34 | igcewieling | We looked into it a few years ago. |
| 23:47.21 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 23:48.42 | Kobaz | oooo |
| 23:48.47 | Kobaz | https://metacpan.org/module/Net::CSTAv3::Client |
| 23:49.24 | Kobaz | i can do some adapting if it's not totally comliant with polycoms |
| 23:49.31 | Kobaz | *compliant |
| 23:52.20 | *** part/#asterisk alhemicar (~alhemicar@178.77.1.189) |