00:03.34 | *** join/#asterisk SuperBawlz (~zbriggs@rrcs-71-43-76-226.se.biz.rr.com) |
00:03.41 | SuperBawlz | anyone home? |
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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 |
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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. |
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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}); |
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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) |
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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 |
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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.. |
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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. |
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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? |
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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 |
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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. |
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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 |
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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 |
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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 |
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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? |
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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. |
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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. |
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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 |
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