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| 01:09.02 | laptopdude91 | Linksys IP phone SPA921 - Any good? |
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| 01:50.48 | tuxd00d | laptopdude91: I have a few of them. Work fine. |
| 01:50.59 | laptopdude91 | Nice thanks |
| 01:51.19 | tuxd00d | Most phones will work. Itâs the extras you have to decide on. |
| 01:51.55 | laptopdude91 | What kind of extras are there? |
| 01:52.49 | Samot | They are unsupported |
| 01:52.49 | tuxd00d | Screen quality, programmable buttons, standard buttons (like hold and transfer). |
| 01:53.21 | tuxd00d | You mean Cisco no longer supports them? |
| 01:54.08 | Samot | Yes. |
| 01:54.25 | tuxd00d | If youâre looking for cheap phones to play with, there are many, such as Yealink. |
| 01:54.52 | tuxd00d | Which still have firmware being developed. |
| 01:56.54 | laptopdude91 | The Linksys phones are $20/piece used |
| 01:58.33 | tuxd00d | A little high for that far out of stupport. |
| 01:58.48 | laptopdude91 | Hm |
| 01:59.32 | laptopdude91 | Is there any reason why I see exten => s, blah blah so much? Does the s signify something? |
| 01:59.41 | tuxd00d | Start |
| 01:59.41 | laptopdude91 | Any reason why not a different letter? |
| 01:59.45 | laptopdude91 | Ah |
| 02:00.32 | tuxd00d | https://wiki.asterisk.org/wiki/display/AST/Special+Dialplan+Extensions |
| 02:01.25 | tuxd00d | âWhen an analog call comes into Asterisk, the call is sent to the s extension. The s extension is also used in macros.â |
| 02:02.07 | laptopdude91 | Thanks! |
| 02:02.14 | laptopdude91 | Asterisk is cool |
| 02:04.08 | laptopdude91 | Sorry for all my dumb questions but I have another one |
| 02:04.13 | laptopdude91 | https://wiki.asterisk.org/wiki/display/AST/Creating+a+Simple+IVR+Menu |
| 02:04.37 | laptopdude91 | This works as intended in all regards except for the fact that it doesn't say the "press 1 or press 2" part |
| 02:04.46 | laptopdude91 | But it says "you entered _" |
| 02:04.56 | laptopdude91 | Wait |
| 02:04.57 | laptopdude91 | Nvm |
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| 02:26.30 | laptopdude91 | Is it possible to have 2 different contexts which are identical except for the ability to place calls to outside the network? |
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| 04:28.34 | tuxd00d | laptopdude91: You can do it that way, or set variables to indicate permissions. |
| 04:29.02 | laptopdude91 | Oooh, permissions sounds much better |
| 04:29.05 | laptopdude91 | I'll look into that, thanks |
| 04:31.24 | tuxd00d | https://wiki.asterisk.org/wiki/display/AST/Conditional+Applications |
| 04:31.44 | tuxd00d | https://wiki.asterisk.org/wiki/display/AST/Setting+and+Substituting+Channel+Variables |
| 04:32.51 | laptopdude91 | Thanks again :) |
| 04:34.31 | laptopdude91 | How would I set a variable depending on the user? |
| 04:40.30 | tuxd00d | ${CHANNEL} contains the sip username but you have to subselect part of it to get just the username. Maybe someone knows a better variable. |
| 04:42.26 | laptopdude91 | Is there any way to set a variable directly in sip.conf? |
| 04:42.43 | tuxd00d | or even better, per the pjsip.conf.sample file, âMYVAR=somevalue ; A user confdigured variableâ |
| 04:43.08 | tuxd00d | So you can set the variable in the user context in pjsip.conf |
| 04:43.21 | tuxd00d | Assuming you are using pjsip. |
| 04:43.28 | laptopdude91 | Unfortunately I'm not |
| 04:45.32 | tuxd00d | Which channel driver are you using? |
| 04:45.35 | tuxd00d | https://wiki.asterisk.org/wiki/display/AST/Channel+Drivers |
| 04:46.02 | laptopdude91 | chan_sip |
| 04:46.34 | tuxd00d | It may be better to learn chan_pjsip as chan_sip is depreciated. |
| 04:46.41 | laptopdude91 | Okay |
| 04:46.48 | laptopdude91 | So if I switch, I no longer need a sip.conf? |
| 04:47.06 | tuxd00d | Correct, which version of Asterisk? |
| 04:47.11 | laptopdude91 | 14 |
| 04:47.34 | tuxd00d | Then yes, pjsip is the obvious better option. |
| 04:48.01 | laptopdude91 | Switching over then |
| 04:48.07 | tuxd00d | While creating users in pjsijp is a little more complicated, the rest is pretty much the same. |
| 04:49.02 | tuxd00d | Using the Wizard is an opton for you, especial starting out when you donât need anything fancy. https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard |
| 04:49.46 | laptopdude91 | Nah, might as well learn from scratch |
| 04:50.08 | tuxd00d | Donât forget to configure the [transport] in pjsip.conf |
| 04:50.49 | tuxd00d | Did you install for source? |
| 04:50.55 | tuxd00d | from source? |
| 04:51.39 | laptopdude91 | Yes |
| 04:51.58 | laptopdude91 | pjsip is scary |
| 04:52.46 | tuxd00d | In the configs/samples directory, youâll find a sample file of each configuration file. |
| 04:53.19 | laptopdude91 | Yeah I'm using that file |
| 04:53.26 | laptopdude91 | Well, looking at it for reference |
| 04:53.41 | tuxd00d | yes, chan_pjsip is intimidating, as it appears more complex than chan_sip. |
| 04:55.50 | laptopdude91 | I'm having trouble understanding what an aor is |
| 04:56.14 | Samot | Address of Record. |
| 04:56.25 | laptopdude91 | What does it do? |
| 04:56.33 | Samot | Chan_PJSIP appears more complex than Chan_SIP because it is. |
| 04:56.39 | drmessano | You can register multiple devices to one extension |
| 04:56.43 | drmessano | Basically |
| 04:56.48 | Samot | It keeps the addresses of the devices for the extension. |
| 04:56.54 | drmessano | They all get a slot, which is an AOR |
| 04:56.58 | Samot | ^^^ |
| 04:57.03 | drmessano | Thats how Asterisk keeps track of them |
| 04:57.17 | drmessano | Thats as simple as I think I can make it |
| 04:57.40 | drmessano | chan_sip was pretty binary.. One device to one account |
| 04:58.00 | drmessano | PJSIP implements AORs, which are awesome |
| 04:58.33 | tuxd00d | laptopdude91: Make sure to read entirely: https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Sections+and+Relationships |
| 04:59.03 | laptopdude91 | So does that mean my phones need static IPs with pjsip? |
| 04:59.13 | Samot | Nope |
| 04:59.26 | tuxd00d | No, Asterisk assigns the IP and port when the phone registers. |
| 04:59.37 | tuxd00d | Not assigns, records. |
| 04:59.41 | Samot | It means that multiple phones can register to the same extension, dynamic or static IP. |
| 05:00.07 | drmessano | laptopdude91: I cant understand how you managed to get this concept so completely wrong |
| 05:00.18 | laptopdude91 | Sorry |
| 05:00.24 | laptopdude91 | It's just they defined an IP in the config |
| 05:00.26 | laptopdude91 | Which is what confused me |
| 05:00.35 | drmessano | ^ Always been there |
| 05:00.37 | Samot | There are multiple samples. |
| 05:00.40 | drmessano | Fixed or Dynamic |
| 05:00.46 | laptopdude91 | Yeah I figured that out haha |
| 05:00.50 | Samot | It shows how to configure a PJSIP device in multiple ways. |
| 05:00.52 | tuxd00d | Many ways to authenticate. |
| 05:00.53 | drmessano | If you want to register, use dynamic |
| 05:01.05 | drmessano | Fixed is for Fixed |
| 05:01.11 | drmessano | Same as chan_sip |
| 05:01.33 | laptopdude91 | Here is my pjsip.conf: https://paste.scd31.com/view/638e55d1 |
| 05:01.45 | laptopdude91 | I'm missing something because it's not working, I can't register |
| 05:02.32 | Samot | Looks fine to me. |
| 05:02.46 | Samot | But there could be various reasons why you can't register. |
| 05:02.51 | Samot | asterisk -rvvvvv |
| 05:02.57 | Samot | pjsip set logger on |
| 05:03.05 | Samot | ^^ Do those from the system CLI |
| 05:03.12 | Samot | Then try to make the device register. |
| 05:03.17 | Samot | ~pb |
| 05:03.17 | infobot | from memory, 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 05:03.21 | Samot | ^^ put the output there. |
| 05:04.13 | laptopdude91 | :5060 isn't listening on the server |
| 05:04.52 | laptopdude91 | No such command 'pjsip set logger on' (type 'core show help pjsip set' for other possible commands) |
| 05:05.33 | Samot | Well then you don't have pjsip loaded. |
| 05:05.42 | laptopdude91 | welp |
| 05:05.46 | tuxd00d | Check modules.conf |
| 05:06.00 | laptopdude91 | [modules] |
| 05:06.00 | laptopdude91 | autoload=yes |
| 05:06.14 | tuxd00d | did you compile it in âmake menuselectâ? |
| 05:06.26 | laptopdude91 | ..no |
| 05:06.28 | laptopdude91 | oops |
| 05:06.39 | laptopdude91 | make menuselect, make, make install? |
| 05:06.39 | Samot | You didn't compile pjsip? |
| 05:06.42 | laptopdude91 | For adding pjsip |
| 05:06.43 | Samot | No. |
| 05:06.49 | laptopdude91 | I just assumed it was defautled |
| 05:07.13 | Samot | https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-bundled |
| 05:07.19 | Samot | Nothing in there indicates that |
| 05:08.05 | laptopdude91 | Whoops |
| 05:09.15 | laptopdude91 | There's an XXX next to it |
| 05:09.26 | tuxd00d | Need prereqs |
| 05:10.48 | tuxd00d | Did you run /contrib/scripts/install_prereq ? |
| 05:11.04 | tuxd00d | It does a fairly good job of getting them. |
| 05:12.02 | laptopdude91 | Doing now |
| 05:12.06 | tuxd00d | Per https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-bundled, âirst, run ./contrib/scripts/install_prereq. Building the bundled pjproject requires the python development libraries which install_prereq has already installed. All you have to do now is add the --with-pjproject-bundled option to your Asterisk ./configure command line and remove any other --with-pjproject option you |
| 05:12.06 | tuxd00d | have specified.â |
| 05:13.28 | tuxd00d | Read this part: https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-bundledUsingtheBundledVersionofpjproject |
| 05:14.03 | laptopdude91 | Thanks |
| 05:14.14 | laptopdude91 | I'm getting there eventually :P |
| 05:15.07 | tuxd00d | This channel doesnât mind helping out, but if itâs already documented, RTFM, as they say :) |
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| 05:33.21 | laptopdude91 | So I decided to install all the extra sounds while I was at it |
| 05:33.27 | laptopdude91 | I may have underestimated their size |
| 05:36.33 | laptopdude91 | Okay so now I'm getting SIP 488 - Not Acceptable Here |
| 05:38.56 | wabbits | pastebin the invite and the 488 response |
| 05:39.27 | wabbits | apologies if I am interloping |
| 05:39.40 | Samot | Well |
| 05:39.49 | Samot | Check your codec settings. |
| 05:39.52 | wabbits | backing off slowly |
| 05:39.55 | laptopdude91 | https://pastebin.ca/3853499 |
| 05:40.37 | Samot | a=rtpmap:110 speex/8000 |
| 05:40.38 | Samot | a=rtpmap:97 iLBC/8000 |
| 05:40.49 | Samot | ^^ That's what you have in your softphone client has the codecs to use |
| 05:40.59 | Samot | Which you do not have allowed in Asterisk. |
| 05:41.10 | laptopdude91 | I fixed it, but I'm sure this is bad practice |
| 05:41.12 | laptopdude91 | allow=all |
| 05:41.19 | Samot | Yes. |
| 05:41.36 | Samot | You should really just allow the codecs you actually want to use |
| 05:41.46 | Samot | And make sure the devices/softphones are using them as well. |
| 05:42.13 | laptopdude91 | Any reason for this? |
| 05:42.41 | Samot | Well |
| 05:42.44 | laptopdude91 | I just want whatever sounds the best, basically |
| 05:42.56 | Samot | ulaw is the general standard |
| 05:43.01 | Samot | speex is garbage |
| 05:43.16 | Samot | iLBC isn't that great. |
| 05:44.09 | wabbits | OPUS, G722, PCMU and PCMA in that order. |
| 05:44.12 | laptopdude91 | Does pjsip.conf have a [general] equivalent? |
| 05:44.16 | laptopdude91 | Where I can define my codecs |
| 05:44.32 | Samot | No. |
| 05:44.34 | Samot | JFC |
| 05:44.42 | Samot | Not every device has opus support yet |
| 05:45.05 | Samot | And since he didn't install PJSIP right the first time around, let's not think Opus is in there. |
| 05:46.09 | Samot | PCMU = North America |
| 05:46.16 | Samot | PMCA = Rest of world |
| 05:46.24 | Samot | That's pretty standard as well. |
| 05:48.49 | Samot | laptopdude91: PCMU (ulaw/g711u) is basically the standard. It's what SIP providers/carriers support to handle calls with the PSTN. |
| 05:48.55 | Samot | Pretty much all devices support it |
| 05:49.07 | laptopdude91 | ulaw is what I've configured the server for now |
| 05:51.12 | Samot | Just make sure it's the first in the order |
| 05:51.19 | Samot | On both Asterisk and on the softphone |
| 06:17.08 | laptopdude91 | Can I get ulaw any better? |
| 06:17.26 | laptopdude91 | Within my network I'd like something decently high quality |
| 06:18.00 | Samot | You can do g722 |
| 06:18.46 | laptopdude91 | Okay |
| 06:18.58 | laptopdude91 | Costs money on zoiper but I'll play with it if I get an IP phone that supports it |
| 06:19.03 | laptopdude91 | Not super high priority |
| 06:19.20 | laptopdude91 | Does the SIP client require port forwarding? |
| 06:46.25 | tuxd00d | laptopdude91: Are you putting Asterisk behind NAT? |
| 06:47.01 | laptopdude91 | Sort of |
| 06:47.10 | laptopdude91 | I may have a SIP client that's on a NAT though |
| 06:47.27 | laptopdude91 | Asterisk will be on a NAT, but on the same NAT as the majority of clients |
| 06:48.05 | tuxd00d | https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT |
| 06:48.42 | tuxd00d | I only deal with clients behind NAT, never a server behind NAT. |
| 06:50.18 | Samot | Its not that hard. |
| 06:50.45 | Samot | Localnet defines the local IP ranges |
| 06:51.14 | Samot | External_media and sugnal |
| 06:51.26 | Samot | Signalling the WAN ip |
| 06:51.40 | laptopdude91 | Does the client need any port forwarding though? |
| 06:51.42 | Samot | And direct_media=no |
| 06:52.16 | Samot | Well you should have 5060 forwarded for the pbx |
| 06:52.32 | Samot | Along with 10000-20000 for rtp |
| 06:52.54 | laptopdude91 | But that's on the server, right |
| 06:53.05 | Samot | Port forwarding needs on the client side are based on the clients network |
| 06:53.54 | tuxd00d | 10000-20000 is the default port range as define in rtp.conf |
| 06:54.30 | laptopdude91 | Does the client have incoming connections? |
| 06:54.44 | Samot | What do you mean? |
| 06:54.52 | tuxd00d | the ports are opened dynamically |
| 06:55.00 | laptopdude91 | On which side though |
| 06:55.19 | drmessano | You dont need ports forwarded for a client |
| 06:55.23 | laptopdude91 | Client is behind a NAT on one internet connection |
| 06:55.24 | tuxd00d | The NAT of the client. |
| 06:55.24 | laptopdude91 | Okay thanks |
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| 08:48.28 | pagios | hi all, http://i.imgur.com/7t1dNRT.png i am trying to configure my extension to be able to make outgoing calls via the FXO port, the extension needs to dial 9 to get the tone and then the number 013466 for example is this configuration correct? |
| 08:53.48 | pagios | ? |
| 08:59.39 | Samot | What is that screenshot? |
| 09:02.26 | Samot | pagios: ^^^ |
| 09:05.41 | pagios | haha |
| 09:05.45 | pagios | dial rules |
| 09:07.08 | Samot | From??? |
| 09:07.30 | Samot | Asterisk doesnt have a GUI |
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| 09:12.49 | pagios | Samot, any website where i can record greeting messages? |
| 09:13.22 | Samot | Dont change the subject |
| 09:16.01 | Samot | Well? |
| 09:20.01 | Samot | Alright then. |
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| 09:32.06 | pagios | Samot, its an atcom device |
| 09:32.43 | Samot | OK and is the 9 a separate deal? |
| 09:32.58 | Samot | Does it want you to press now and then digits to match against |
| 09:33.07 | Samot | or is it matching against the 9+ digits? |
| 09:33.57 | pagios | i want to press0 and then dial the pstn numebr |
| 09:34.14 | Samot | That's not what you said originally |
| 09:35.16 | Samot | _XXXXXX <- That's going to match any 6 digits |
| 09:35.20 | Samot | That's it |
| 09:35.37 | Samot | Could start with a 9 or a 0 or anything since it's X |
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| 09:50.20 | pagios | Samot, it works now thanks |
| 09:50.41 | pagios | question, if i want to integrate with skype connect to redirect incoming PSTN to SIP i need to pay any monthly fees to skype? |
| 09:50.45 | pagios | or the gateway is free |
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| 12:32.42 | pagios | is there a way to make users trying to call me on skype , redirected into asterisk trunk?\ |
| 12:32.48 | pagios | they are calling not by number |
| 12:32.57 | pagios | they are calling by name (soft phone) |
| 12:33.09 | *** join/#asterisk Eloy (~Eloy@5.149.168.66) |
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| 12:40.33 | pagios | CAN I MAKE incoming pstn calls routed into my SKYPE account so that my skype rings? |
| 12:40.41 | pagios | is this possible or aam i wasting time trying |
| 12:41.27 | [TK]D-Fender | "skype sip" <_ GOOGLE IT. |
| 12:41.34 | [TK]D-Fender | Do you see anything? |
| 12:43.11 | pagios | [TK]D-Fender, i DID GOOGLE IT |
| 12:43.20 | [TK]D-Fender | And? |
| 12:43.29 | pagios | it works on the PSTN line that you buy from skype |
| 12:43.40 | pagios | i need to MYOWNPSTNLINE->SKYPEUSER |
| 12:43.50 | pagios | not skypepstnline->skypeuser |
| 12:44.20 | pagios | so the rule should be incoming PSTN line -> routed over skype trunk-> skypeusername rings |
| 12:47.12 | [TK]D-Fender | chan_motif.conf <---- Looked at yet? |
| 12:47.58 | [TK]D-Fender | Though Skype is in the middle fo killing off everything in lieu of Hangouts |
| 12:47.58 | pagios | [TK]D-Fender, me? |
| 12:48.15 | [TK]D-Fender | WHO ELSE IS TALKING? |
| 12:48.47 | eric_hill | grabs a popcorn |
| 12:49.03 | [TK]D-Fender | If you Google'd "asterisk skype" you should have found this long ago |
| 12:49.18 | *** join/#asterisk drathir (~kamiljk8@unaffiliated/drathir) |
| 12:49.33 | pagios | OMG |
| 12:49.43 | pagios | i am using ATCOM appliance |
| 12:50.27 | pagios | and they dont mention it, http://blog.ipchitchat.co.uk/wp-content/uploads/2010/11/IP0x_skype.pdf this is why ia m asking an asterisk quesiton |
| 12:53.34 | [TK]D-Fender | And what version is that thing running? |
| 12:54.32 | [TK]D-Fender | that thing is an ancient piece of garbage. |
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| 14:36.38 | pagios | hi, do i need a real ip on asterisk for skype to be able to forward sip calls into my pbx? |
| 14:44.29 | Samot | Skype forwards a call like a regular call forwarding.. |
| 14:49.29 | [TK]D-Fender | What's a "real IP"? |
| 14:49.52 | [TK]D-Fender | What do their instructions say? |
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| 15:15.10 | jes__ | Hello all, I am attempting to test asterisk a bit for evaluation. I am using Asterisk 13.13, Rhel 7, 8 core Xeion E5, 16 GB RAM. I have two scenarios that I was hoping to get some feedback on. First is an extension that is Answer, PlayBack, MusicOnHold, Hangup. I run Sipp on another server with a scenario that is Invite, Wait for Answer, Ack, Pause 8 secound, Bye. The pause is long enough for the server to play the audio and reach Moh. Sipp starts timing o |
| 15:15.50 | jes__ | Does this seem about right to you? |
| 15:17.24 | jes__ | CPU usage hits around 40-50% |
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| 15:20.13 | pagios | [TK]D-Fender, i did an incoming rule where i DIAL a number via the SKYPE trunk, is there a way to dial my skype account instead of a number? this way all pstn incoming calls would be forwarded to my skype username and i can answer over ot |
| 15:20.32 | pagios | i want to forward to skype username and not a number |
| 15:20.51 | [TK]D-Fender | Did you try? |
| 15:20.56 | [TK]D-Fender | Got something to show us? |
| 15:20.57 | pagios | yea it doesnot work |
| 15:21.07 | [TK]D-Fender | If you're using that stupid appliance, it's not supported |
| 15:21.17 | [TK]D-Fender | It's GUI, that Asterisk version, none of it |
| 15:21.42 | pagios | it is based on asterisk yea |
| 15:21.46 | [TK]D-Fender | means nothng |
| 15:21.46 | pagios | in regular asterisk does it work? |
| 15:21.59 | [TK]D-Fender | their shit GUI controls everythiing and it's running an ancient version of * that is not supported |
| 15:22.04 | pagios | i can edit asterisk files intht appliance |
| 15:22.06 | [TK]D-Fender | Just the same as Windows 3/11 |
| 15:22.20 | pagios | Ok what do you recommend |
| 15:22.30 | [TK]D-Fender | Get a supported version |
| 15:22.41 | [TK]D-Fender | And learn it |
| 15:22.42 | pagios | which version onwahrd? |
| 15:22.46 | [TK]D-Fender | 13 |
| 15:23.38 | pagios | http://www.atcom.cn/dl_ip02.html thats all i see hehe [TK]D-Fender |
| 15:24.16 | [TK]D-Fender | That is meaningless |
| 15:24.20 | Samot | What Skype trunk? |
| 15:24.48 | pagios | Skype trunk outgoing is working fine |
| 15:24.58 | [TK]D-Fender | Samot, He's using an ancient piece of shit appliance with a GUI to match |
| 15:25.00 | pagios | i tried to put a rule to dial into the trunk and it works |
| 15:25.12 | [TK]D-Fender | pagios, We don't support that thing |
| 15:25.28 | pagios | so i can pick my phone, dial 6 and then a number and that would go into skype trunk it works |
| 15:25.43 | pagios | but my other setup to forwatd PSTN to skype username does not work |
| 15:25.58 | Samot | Well, there is also the fact that Skype doesn't support it |
| 15:26.04 | Samot | But that's a little thing. |
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| 15:28.46 | Samot | Skype for SIP no longer exists. |
| 15:28.48 | pagios | Samot, the whole idea is that i am not in the office sometimes and i want ppl calling into the pstn office line to be forwarded to my phone which has sip |
| 15:28.55 | Samot | Skype Connect does not support Asterisk. |
| 15:28.57 | pagios | how can i achieve that |
| 15:29.13 | pagios | i dont mind if it is something other than skype |
| 15:29.20 | Samot | You call forward like any other call. |
| 15:29.31 | pagios | i need to call forward over SIP |
| 15:29.36 | Samot | From? |
| 15:29.39 | pagios | dont want to be charged for incoming calls |
| 15:29.41 | Samot | Call forward from where? |
| 15:29.41 | pagios | from PSTN |
| 15:29.52 | Samot | You can;t |
| 15:29.55 | pagios | so client -> PSTN -> SIP-> my phone |
| 15:30.01 | Samot | No. |
| 15:30.04 | pagios | why? |
| 15:30.13 | [TK]D-Fender | <pagios> so client -> PSTN -> SIP-> my phone <- where the fuck is SKYPE in that picture? |
| 15:30.13 | Samot | Only if that call is being routed to your PBX via your SIP provider. |
| 15:30.35 | pagios | [TK]D-Fender, SIP can be the skype gateway |
| 15:30.37 | Samot | Client -> Client Carrier -> PSTN -> Destination Carrier -> Destination User |
| 15:30.41 | Samot | ^^^ That's how a call works |
| 15:30.43 | [TK]D-Fender | Don't jsut say "SIP |
| 15:30.53 | [TK]D-Fender | <PROTECTED> |
| 15:31.17 | pagios | ok DEStination carrier works over sip here right |
| 15:31.22 | Samot | OK |
| 15:31.29 | Samot | So it would send the call to your PBX like normal |
| 15:31.35 | pagios | and destination user would be my sip account |
| 15:31.41 | Samot | If you're out of the office, you forward calls to your extension. |
| 15:31.48 | Samot | What SIP account?! |
| 15:31.55 | pagios | one second |
| 15:32.01 | pagios | let me rephrase |
| 15:32.15 | Samot | No. |
| 15:32.17 | Samot | Let me |
| 15:32.26 | Samot | If I call your business number, where do it go? |
| 15:32.30 | Samot | To your PBX? |
| 15:32.45 | Samot | This Atcom device? |
| 15:32.55 | pagios | yes |
| 15:32.58 | pagios | via FXO |
| 15:32.59 | [TK]D-Fender | He's mixing topics, directions & technologies like a a friggen smootie |
| 15:33.09 | Samot | OK |
| 15:33.15 | Samot | So your business number is on a POTS line. |
| 15:33.20 | pagios | yes |
| 15:33.23 | Samot | So standard phone service. |
| 15:33.27 | pagios | correct |
| 15:33.31 | Samot | OK so when that call his your Atcom, then what? |
| 15:33.36 | Samot | It rings a phone? |
| 15:33.41 | pagios | no |
| 15:33.53 | pagios | i want it to ring a softphone on my mobile phone whereever i am |
| 15:34.03 | pagios | even if in office |
| 15:34.05 | Samot | OK |
| 15:34.08 | Samot | So what is the issue? |
| 15:34.21 | pagios | i am trying to see how i can achieve this |
| 15:34.28 | Samot | You send the call to the extension you're registered with on the softphone |
| 15:34.32 | pagios | skype being that softphone doesnt seem to work |
| 15:34.34 | Samot | Well that's in the AtCom |
| 15:34.36 | Samot | We can't help |
| 15:34.45 | pagios | fuck atcom |
| 15:34.54 | Samot | This is a proprietary system that uses Asterisk |
| 15:34.57 | pagios | i want to understand what i can use if i use asterisk |
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| 15:35.07 | Samot | You can't do it |
| 15:35.10 | Samot | Not with the AtCom |
| 15:35.28 | pagios | i just wanna do it , i dont care if atcom or standard asterisk |
| 15:35.42 | Samot | 11:34:28 AM S<Samot> You send the call to the extension you're registered with on the softphone |
| 15:36.00 | pagios | so i need a sip provider account |
| 15:36.06 | Samot | Huh? |
| 15:36.16 | Samot | Doesn't the AtCom support SIP extensions? |
| 15:36.23 | pagios | it does |
| 15:36.27 | Samot | OK |
| 15:36.35 | Samot | So you make a SIP extension. |
| 15:36.40 | Samot | Just like you would on Asterisk. |
| 15:36.48 | Samot | You register your softphone to that extension. |
| 15:36.49 | Samot | Done. |
| 15:36.58 | Samot | Now you have your phone and your softphone. |
| 15:37.09 | pagios | that would work inside the office |
| 15:37.11 | Samot | If calls to the phone extension aren't answered or whatever, send them to the other extension. |
| 15:37.13 | pagios | on the same network |
| 15:37.30 | Samot | Well then you need to setup your AtCom to allow remote registrations |
| 15:37.37 | Samot | If you can't do that |
| 15:37.39 | pagios | then i need real ips |
| 15:37.43 | Samot | Then throw it out and get a real PBX |
| 15:37.44 | Samot | No |
| 15:37.50 | Samot | You already have a real IP |
| 15:37.55 | Samot | It's called your WAN IP |
| 15:38.00 | Samot | Don't you have Internet? |
| 15:38.05 | pagios | its natted |
| 15:38.13 | pagios | i need to reverse nat to do what you just described |
| 15:38.15 | Samot | It's going to be NATted |
| 15:38.24 | Samot | Because your system is behind NAT |
| 15:38.46 | pagios | yes so to allow my extension to work outside the office i need a reverse nat |
| 15:38.51 | [TK]D-Fender | we're skipping the final point of failure here |
| 15:38.57 | Samot | "Reverse NAT"? |
| 15:38.58 | Samot | WTF? |
| 15:39.07 | pagios | reverse port forwarding |
| 15:39.16 | Samot | Uhm. |
| 15:39.24 | voipmonk | moonwalks |
| 15:39.33 | Samot | You have a pfSense firewall don't you? |
| 15:39.45 | Samot | Because this sounds like some pfSense crap. |
| 15:39.46 | pagios | no |
| 15:39.50 | Samot | "reverse port forwarding" |
| 15:40.22 | pagios | the softphone is reached if i am inside the office , how do you reach the pbx if you are outside the office |
| 15:40.44 | Samot | You need to do some research |
| 15:40.50 | [TK]D-Fender | The word "reverse" does not come into this |
| 15:40.57 | Samot | I can't explain Networking 101 to you for this. |
| 15:41.02 | [TK]D-Fender | it's PORT FORWARDING |
| 15:41.13 | [TK]D-Fender | Yes you forward the OUTSIDE shit from your router to your PBX |
| 15:41.22 | pagios | [TK]D-Fender, exactly, |
| 15:41.27 | [TK]D-Fender | Don't say "reverse" |
| 15:41.30 | pagios | and i dont have access to forward this shit |
| 15:41.32 | [TK]D-Fender | So fine. |
| 15:41.36 | [TK]D-Fender | you do that |
| 15:41.40 | Samot | Then you're screwed. |
| 15:41.51 | pagios | https://www.google.com.lb/search?q=reverse+nat&oq=reverse+nat&aqs=chrome..69i57j0l5.1979j0j7&sourceid=chrome&ie=UTF-8 |
| 15:41.55 | Samot | You can't access your router/firewall then you can't setup remote access correctly. |
| 15:41.59 | [TK]D-Fender | You set up * so it knows what your WAN IP is and treats your calls appropriately |
| 15:42.16 | pagios | Samot, exactly |
| 15:42.24 | pagios | so the sip provide can help in this case |
| 15:42.29 | pagios | coz the sip provider is in the cloud |
| 15:42.45 | [TK]D-Fender | pagios, Not a lot to do to save you from a system that is fucking you. |
| 15:42.52 | pagios | and if i connect my pbx to it my pbx would act as a client and that sip provider can take th request and send them inside that firewall |
| 15:43.00 | Samot | pagios: |
| 15:43.08 | Samot | To connect your PBX in the cloud to your PBX in the office.. |
| 15:43.14 | Samot | YOU NEED ROUTER ACCESS |
| 15:43.20 | Samot | Because you still need NAT rules |
| 15:43.37 | Samot | Because, guess what, that office PBX is still behind NAT |
| 15:43.38 | [TK]D-Fender | otherwise audio and calls aren't getting in and you're fucked |
| 15:44.04 | pagios | it is behind nat but acting as a client in this case |
| 15:44.07 | pagios | not as a server |
| 15:44.11 | Samot | Also "1-to-1 or 1-to-many" NAT is some Cisco phrasing. |
| 15:44.16 | pagios | so it is server cloud sending to the pbx in the office |
| 15:44.26 | pagios | so you dont need nat rules |
| 15:44.34 | Samot | "Reverse NAT" which they describe as the WAN IP to a Private IP is called PORT FORWARDING |
| 15:44.49 | pagios | call it whatever |
| 15:44.55 | Samot | Sure. |
| 15:45.10 | Samot | You don't have control of your router/firewall it's all moot anyways |
| 15:45.19 | Samot | Because you can't setup the needed rules |
| 15:47.24 | pagios | what i will do is deploy asterisk in the cloud as a sip server with extensions loaded. have my office pbx connect to that cloud pbx via sip and route to the extension in the cloud |
| 15:47.29 | pagios | i dont need nat rules for this |
| 15:47.46 | Samot | OK. |
| 15:47.48 | pagios | my phone would then be provisioned with the cloud extension |
| 15:47.53 | Samot | If you dont think so. |
| 15:48.15 | pagios | Samot, my fucking pbx is now in the office and it is routing to skype connect correcly with no nat rules |
| 15:48.21 | pagios | replace skype by my asterisk cloud |
| 15:48.22 | pagios | same thing |
| 15:48.47 | Samot | You're using Skype for OUTBOUND |
| 15:48.53 | pagios | correct |
| 15:48.55 | Samot | The incoming calls go over your POTS line |
| 15:49.08 | Samot | Incoming calls over SIP are different than outgoing calls over SIP in regards to NAT |
| 15:49.24 | pagios | incoming calls go over potsd -> go into atcom-> atcom sip it into asterisk cloud -> into softphone extension |
| 15:49.41 | pagios | i dont have incoming calls over sip in the office |
| 15:49.55 | pagios | i have only incoming pstn |
| 15:50.25 | pagios | where do you see incoming calls over sip in my case/ |
| 15:50.49 | Samot | Go for it |
| 15:50.54 | Samot | You seem to have the answers. |
| 15:51.00 | Samot | You seem to understand what's needed |
| 15:51.00 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 15:51.02 | Samot | Go to town on it. |
| 15:51.26 | Samot | When you come back with audio issues and we say "Fix your firewall" and you say "I don't have access" |
| 15:51.31 | Samot | That's it. Done. |
| 15:52.00 | pagios | Samot, how about i create a vpn then |
| 15:52.19 | pagios | and tunnel all that over between the office pbx and the cloud one |
| 15:52.56 | Samot | Well, you need access to your router/firewall |
| 15:53.08 | Samot | So throw that idea out the window. |
| 15:53.11 | pagios | i can run a vpn server on port 80 |
| 15:54.50 | Samot | OK |
| 15:54.53 | Samot | Like I said.. |
| 15:55.00 | Samot | You seem to have it figured out, go to town. |
| 15:57.00 | Samot | How you want to do this, to me, is 100% hackish and not the right way. |
| 15:57.36 | [TK]D-Fender | Have we done away with SKYPE in this situation entirely? |
| 15:58.02 | Samot | Not really sure. |
| 15:58.21 | Samot | Since the cloud PBX so far is to facilitate the registration of a softphone.. |
| 15:58.36 | Samot | Nothing has been said on how it would connect to the PSTN for outbound calls. |
| 15:58.44 | Samot | Or if it would be used for that |
| 15:59.33 | pagios | Samot, i dont need the softphone toconnect to pstn |
| 15:59.40 | Samot | I know. |
| 15:59.48 | pagios | i just need incoming pstn to go to softphone thats it :) |
| 15:59.50 | Samot | But you use Skype for outbound. |
| 15:59.51 | [TK]D-Fender | ATCOM on that LAN side with FXO's |
| 15:59.56 | Samot | Are you still going to do that? |
| 16:00.09 | Samot | How would the softphone make an outbound call?! |
| 16:00.11 | pagios | Samot, i will get rid of skype |
| 16:00.13 | [TK]D-Fender | SKYPE was to a SOFT CLIENT |
| 16:00.28 | pagios | the softphone doesnt make outbound calls |
| 16:00.31 | pagios | it only accept calls |
| 16:00.44 | pagios | outbound calls are made from the office via FXO ports |
| 16:01.13 | Samot | So when you are out of the office. |
| 16:01.16 | Samot | You can receive calls.. |
| 16:01.22 | pagios | right! |
| 16:01.25 | pagios | thats all i want |
| 16:01.27 | Samot | But you won't be able to do follow ups or return calls. |
| 16:01.28 | *** join/#asterisk tzafrir (~tzafrir@206.167.44.205) |
| 16:01.35 | pagios | correct |
| 16:01.43 | Samot | OK |
| 16:01.46 | pagios | i can call over my phone regular GSM for that |
| 16:02.25 | pagios | Samot, are there like sip providers that can replace the hassle of having my own asterisk in the cloud |
| 16:02.44 | wabbits | yes |
| 16:02.46 | pagios | i would then integrate the office pbx with them |
| 16:02.46 | *** join/#asterisk miralin (~Thunderbi@91.237.94.2) |
| 16:02.49 | drmessano | Flowroute.com |
| 16:02.52 | pagios | instead of getting a hosting and stuff |
| 16:02.53 | drmessano | Amongst many |
| 16:03.03 | drmessano | or just edit the MySQL DB directly |
| 16:03.06 | drmessano | ^ ignore that |
| 16:03.18 | drmessano | flowroute.com is a good choice |
| 16:03.29 | Samot | Stop. |
| 16:03.31 | Samot | JFC. |
| 16:03.44 | Samot | Your PBX in the office is an Atcom FXO device. |
| 16:03.54 | Samot | You do not have access to your firewall/router |
| 16:04.14 | Samot | How integrating a SIP provider into your office PBX become a thing now? |
| 16:04.22 | Samot | Along with being able to register your softphone? |
| 16:04.27 | pagios | yea |
| 16:04.41 | pagios | i would register my softphone with the provider instead of my asterisk cloud |
| 16:04.51 | Samot | Oh FFS. |
| 16:05.02 | Samot | Let me know when you are serious about a solution for this. |
| 16:06.13 | Samot | How would your calls get from the Atcom to the SIP provider account your phone is registered to? |
| 16:06.20 | Samot | They wouldn't. |
| 16:06.26 | drmessano | Please stop calling it a fucking cloud |
| 16:06.31 | drmessano | Its painful |
| 16:06.53 | Samot | So far the options have been a hosted VM instance of Asterisk or a SIP provider account.. |
| 16:07.04 | Samot | That would ONLY facilitate the softphone registration. |
| 16:07.17 | drmessano | So you have an ATCOM |
| 16:07.19 | Samot | How are you getting the calls from the Atcom in your office to either of those? |
| 16:07.41 | Samot | Incoming Call from PSTN over POTS to Atcom FXO device. |
| 16:07.49 | wabbits | pagios hire a consultant |
| 16:08.33 | drmessano | pagios: This is real easy |
| 16:09.22 | wabbits | ^ |
| 16:09.27 | drmessano | Set up an Asterisk VM somewhere |
| 16:09.41 | drmessano | Register your softphone to it |
| 16:09.46 | drmessano | Register the ATCOM to it |
| 16:09.50 | drmessano | Wirenut |
| 16:09.51 | drmessano | Done |
| 16:10.18 | Samot | And hope you don't need to make any firewall/NAT rule changes. |
| 16:10.42 | drmessano | Well, the ATCOM will register the Asterisk VPS |
| 16:10.56 | drmessano | So mostly likely not |
| 16:10.57 | Samot | Yeah, it should take care of it |
| 16:11.06 | Samot | But who knows what is running. |
| 16:11.10 | drmessano | You'll need ports open in the VM, of course |
| 16:11.36 | drmessano | If you DO have NAT issues |
| 16:11.39 | drmessano | Get another ATA |
| 16:11.47 | drmessano | Because you're stuck with the ATCOM |
| 16:11.55 | drmessano | As far as Asterisk newness |
| 16:12.59 | drmessano | But I could register a SPA-3102 across our convoluted WAN to my Asterisk box at home, and had no problems |
| 16:13.16 | drmessano | But the Linksys/Sipura stuff always just "works" |
| 16:14.47 | Samot | What kind of router do you have, pagios? |
| 16:15.26 | drmessano | pagios: Please do not DM |
| 16:15.48 | drmessano | Beating are public here, thanks |
| 16:15.52 | drmessano | BeatingS are public here, thanks |
| 16:15.54 | pagios | the vm you are talking about is hosted in the cloud |
| 16:16.01 | drmessano | Dude stop |
| 16:16.01 | pagios | beatings? |
| 16:16.20 | drmessano | Let me go over this one more time |
| 16:16.43 | drmessano | ATCOM --Register--> Asterisk VM <--Register-- Softphone |
| 16:16.44 | drmessano | Done |
| 16:17.07 | pagios | ^^ where are your stuff hosted |
| 16:18.11 | pagios | all inside the office? |
| 16:18.43 | drmessano | No |
| 16:18.47 | drmessano | Hosted VM |
| 16:18.53 | drmessano | Linode |
| 16:19.00 | pagios | so its called fucking cloud |
| 16:19.14 | Samot | Wow. |
| 16:19.14 | drmessano | JFC |
| 16:19.16 | Samot | Really? |
| 16:19.22 | drmessano | Its someone elses fucking computer |
| 16:19.25 | drmessano | Not a fucking cloud |
| 16:19.39 | drmessano | Calling it "Asterisk Cloud" over and over again sounds retarded |
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| 16:19.41 | *** mode/#asterisk [+o bford] by ChanServ |
| 16:20.50 | drmessano | Sounds like arguing is more important than actually getting this working |
| 16:21.32 | pagios | that was the scenario i was talking about |
| 16:21.48 | pagios | to have my atcom-> into a VPS and a softphone register wit hthe VPS |
| 16:21.51 | Samot | Yes but "cloud" is a broad term. |
| 16:22.01 | pagios | and have on atcom a rule to fotward pstn to the VPS extension |
| 16:22.02 | Samot | It can refer to any type of network. |
| 16:22.04 | drmessano | pagios: So whats the issue? |
| 16:22.04 | pagios | omg |
| 16:22.18 | Samot | Well, you're insisting on using the term. |
| 16:22.20 | Samot | Understand it. |
| 16:22.24 | pagios | we were arguing ifwe need nat rules and i said we dont need ones on the atcom site |
| 16:23.03 | pagios | lol ok VPS , word of the day cool cool |
| 16:23.35 | Samot | Well.. |
| 16:23.39 | Samot | Here's the thing... |
| 16:23.51 | Samot | Your Atcom supports SIP registrations for extensions. |
| 16:24.01 | Samot | You really could register your softphone to the Atcom. |
| 16:24.03 | pagios | drmessano, my second argument was that instead of having a self hosted VPS with asterisk running on it , i was asking if you can have asterisk-sip-oroviders instead of the vps |
| 16:24.03 | Samot | Be done with it. |
| 16:24.13 | pagios | Samot, no |
| 16:24.19 | pagios | coz you can use it only in the office then |
| 16:24.21 | wabbits | pagios I think you should come back later with doughnuts. |
| 16:24.28 | Samot | pagios: No. |
| 16:24.31 | pagios | Samot, yes |
| 16:24.33 | Samot | You could use it remotely. |
| 16:24.35 | Samot | FFS. |
| 16:24.35 | pagios | no |
| 16:24.39 | Samot | It's done all the time. |
| 16:24.45 | pagios | coz you need port forwarding for this |
| 16:24.52 | pagios | and i dont have access to that |
| 16:24.53 | Samot | drmessano: Does your softphone register via LAN and then WAN when you leave? |
| 16:24.59 | Samot | RIGHT |
| 16:25.12 | Samot | So we're hacking this. |
| 16:25.14 | pagios | i dont have port forwarding access, for the 100 times |
| 16:25.27 | Samot | The answer is to get another PBX to make this happen |
| 16:25.39 | pagios | Samot, the other PBX is that VPS |
| 16:25.42 | Samot | One that is not on your network. |
| 16:25.48 | Samot | Sure. |
| 16:25.51 | pagios | so my question is cn that other PBX be a sip-provider |
| 16:25.58 | pagios | where i subscribe to services |
| 16:26.02 | pagios | instead of deploying my own |
| 16:26.02 | Samot | Sure. |
| 16:26.11 | pagios | great now we are on the same page |
| 16:26.14 | pagios | what companies? |
| 16:26.20 | Samot | flowroute.com |
| 16:26.23 | pagios | ok |
| 16:26.26 | Samot | You can register that PBX to Flowroute |
| 16:26.30 | pagios | so atcom would integratewith them |
| 16:26.34 | Samot | Who? |
| 16:26.38 | Samot | Flowroute? |
| 16:26.45 | pagios | ATCOM -> flowroute |
| 16:26.48 | Samot | OK |
| 16:26.54 | Samot | How does that solve your softphone issue? |
| 16:27.05 | pagios | my softphone registers with flowroute! |
| 16:27.20 | Samot | So your Atcom registers to Flowroute |
| 16:27.23 | pagios | and my atcom registers with flowroute |
| 16:27.28 | Samot | Your softphone registers to flowroute |
| 16:27.37 | pagios | and i can route my pstn to the flowroute extension |
| 16:27.38 | Samot | Flowroute will treat those as two entities. |
| 16:27.43 | pagios | yes! |
| 16:27.58 | Samot | So there could be a charge. |
| 16:28.12 | pagios | yea but low |
| 16:28.27 | pagios | hosting my vps will have a charge too |
| 16:28.29 | Samot | Uhm. |
| 16:28.38 | Samot | Account 1 has to call account 2 |
| 16:28.38 | pagios | https://www.flowroute.com/pricing/ |
| 16:28.42 | Samot | Account 1 is outbound |
| 16:28.45 | Samot | Account 2 is inbound |
| 16:28.47 | pagios | i think i need the inbound calling in my case |
| 16:28.48 | Samot | You're paying twice. |
| 16:28.51 | Samot | NO |
| 16:28.58 | Samot | To make the Atcom call your softphone |
| 16:29.07 | Samot | That is one SIP account calling another SIP account |
| 16:29.12 | pagios | right |
| 16:29.16 | Samot | You have to call a DID |
| 16:29.23 | Samot | The softphone needs a DID to accept CALLS |
| 16:29.45 | pagios | ok so? |
| 16:30.06 | Samot | So your Atcom trunk will be making an Outbound call as far as Flowroute is concerned. |
| 16:30.14 | Samot | Which is calling the DID to your softphone. |
| 16:30.20 | Samot | Which is an incoming call |
| 16:30.23 | pagios | yes |
| 16:30.28 | Samot | So now your paying for an outbound call and and INBOUND call |
| 16:30.34 | Samot | $0.0098 on one |
| 16:30.40 | Samot | And $0.012 on the other |
| 16:30.44 | Samot | So "low" |
| 16:31.05 | [TK]D-Fender | Why are we looking at 2 ITSP accounts here exactly? |
| 16:31.11 | Samot | Don't know |
| 16:31.19 | [TK]D-Fender | Curious how this balooned into a stupid mess? |
| 16:31.32 | Samot | The answer was given |
| 16:31.33 | [TK]D-Fender | (stupid-ER) |
| 16:31.38 | Samot | Asterisk VM |
| 16:31.46 | Samot | Atcom registers to Asterisk |
| 16:31.51 | Samot | Softphone registers to Asterisk |
| 16:31.52 | Samot | Done. |
| 16:31.55 | [TK]D-Fender | Why is there an ITSP at all |
| 16:32.02 | Samot | Don't know, he asked. |
| 16:32.03 | [TK]D-Fender | We had FXO's already |
| 16:32.16 | Samot | He asked if he could trunk the Asterisk VM to a provider. |
| 16:32.18 | pagios | i am studying all options |
| 16:32.18 | Samot | We said "yes" |
| 16:32.46 | Samot | pagios: What kind of router do you have? |
| 16:32.49 | [TK]D-Fender | And what "option" does that represent? |
| 16:32.51 | Samot | And why don't you have access? |
| 16:32.57 | pagios | Samot, paloalto |
| 16:33.25 | pagios | anyways |
| 16:33.32 | Samot | Well |
| 16:33.54 | Samot | The correct option is to do this with the on-premises PBX and not have to add a second PBX |
| 16:34.01 | Samot | For the record. |
| 16:34.25 | pagios | i thought initially the skype trunk could be the DID |
| 16:34.41 | Samot | Skype for SIP is dead |
| 16:34.45 | Samot | It's Skype Connect now |
| 16:34.50 | Samot | It does not work with Asterisk. |
| 16:34.57 | pagios | google hangouts work? |
| 16:35.00 | Samot | Asterisk is not a certified PBX with Skye Connect. |
| 16:35.01 | Samot | No |
| 16:35.03 | pagios | anything else that ic an leverage? |
| 16:35.25 | [TK]D-Fender | For what? |
| 16:35.30 | Samot | Hangouts does not have third party integration. |
| 16:35.44 | pagios | Samot, i see. anything else that has? |
| 16:35.51 | Samot | For what? |
| 16:35.59 | Samot | Leverage for what? |
| 16:36.18 | pagios | something like skype connect that would work with asterisk |
| 16:36.38 | Samot | For what? |
| 16:36.38 | pagios | that way i can get it for free instead of deploying a vps |
| 16:36.47 | Samot | OK. |
| 16:36.54 | Samot | You are clearly not grasping this. |
| 16:38.15 | Samot | You want to be able to "call forward" calls from your office PBX to your cell phone. |
| 16:38.19 | Samot | That's the goal. |
| 16:38.33 | Samot | Without having to either pay or pay much for it. |
| 16:39.08 | Samot | Have an ITSP/Provider service in between is the equivalent of a real call forwarding. |
| 16:39.39 | Samot | You really want your cell phone to have a softclient that registers to your PBX as an extension. |
| 16:39.58 | Samot | So the calls are extension to extension and thus not "forwarded" over the PSTN |
| 16:41.19 | Samot | Since you cannot do this the right way for whatever reasons... |
| 16:41.36 | Samot | You need a work around, which in this case is another PBX or B2BUA system. |
| 16:41.52 | Samot | That is not on your office network. |
| 16:43.21 | pagios | btw fot the port forward how many ports do you need? |
| 16:43.32 | wabbits | pools of liquid metal are aggregating |
| 16:44.03 | Samot | You need 5060 on UDP for signalling and 10000-20000 on UDP for RTP/media |
| 16:45.36 | pagios | no TCP needed i see |
| 16:45.50 | Samot | You can do TCP if you'd like. |
| 16:45.57 | Samot | But that is only for signalling. |
| 16:46.02 | Samot | RTP is always UDP. |
| 16:46.06 | pagios | ok |
| 16:46.09 | pagios | thanks |
| 16:48.02 | Samot | You should also make sure that SIP ALG is disabled on the router. |
| 16:48.13 | Samot | It can cause some havoc if it is enabled. |
| 16:49.21 | Samot | If the ATCOM allows it on the extension in it's setting, make sure it has Qualify/Keep-Alive set to Yes. |
| 16:49.36 | Samot | So the ATCOM will try and keep the NAT hole open. |
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| 17:00.45 | drmessano | pagios: This is a mobile softphone? |
| 17:04.19 | pagios | yes |
| 17:04.29 | drmessano | My suggestion |
| 17:04.39 | drmessano | Get this working with UDP |
| 17:04.43 | drmessano | then switch to TCP |
| 17:05.08 | drmessano | TCP is a fuckton better on battery life |
| 17:10.40 | wabbits | pagios iphone ? |
| 17:13.55 | pagios | anything |
| 17:14.13 | pagios | btw for the call greeting .wav file what specs should it have? 8bit? |
| 17:14.39 | wabbits | iphone will be yet another problem :) |
| 17:14.50 | drmessano | No it won't |
| 17:14.53 | drmessano | Use Bria |
| 17:17.26 | wabbits | So who is going to wake up the b |
| 17:17.34 | wabbits | ria client to recieve calls? |
| 17:18.01 | drmessano | The Bria service that's integrated into the client |
| 17:18.21 | wabbits | but it will be asleep |
| 17:18.22 | drmessano | As of 5.0 |
| 17:18.31 | drmessano | No |
| 17:18.36 | drmessano | We had this discussion before |
| 17:18.46 | drmessano | Bria 4.x backgrounded PROPERLY on IOS 10.3 |
| 17:19.00 | drmessano | Im sorry Zoiper decided not to update |
| 17:19.05 | drmessano | But Counterpath did |
| 17:19.16 | drmessano | Now with 5.x they have a Push service |
| 17:19.28 | wabbits | yes and somebody has to initiate the push |
| 17:19.31 | drmessano | Which completely sleeps the client until a push, should you choose to use it |
| 17:19.37 | drmessano | Their service does |
| 17:19.49 | wabbits | how does their service know to do this? |
| 17:20.31 | drmessano | You trigger it in the dialplan |
| 17:20.45 | drmessano | They have a simple API for it |
| 17:20.46 | wabbits | this is the yet another problem that I referred to |
| 17:20.52 | drmessano | However |
| 17:21.02 | drmessano | I background on 10.3 just fine with 4.x and 5.x |
| 17:21.12 | drmessano | Haven't missed a call |
| 17:21.22 | drmessano | The Push service is more brute force |
| 17:22.04 | drmessano | You keep missing the part where Counterpath updated Bria with proper 10.3 background support |
| 17:22.16 | wabbits | nope |
| 17:22.22 | wabbits | I don't think I do |
| 17:22.33 | drmessano | Then you're just trolling |
| 17:22.39 | drmessano | I dont have time for that |
| 17:22.46 | wabbits | suit yourself |
| 17:23.01 | Samot | None of my clients have an issue with Bria. |
| 17:23.03 | drmessano | Doesn't change that Bria works under 10.3 |
| 17:23.11 | drmessano | They updated |
| 17:23.17 | Samot | They mostly use iPhones. |
| 17:23.18 | drmessano | It resolved the background issues |
| 17:23.25 | drmessano | The problem with Zoiper is ZOIPER |
| 17:23.28 | drmessano | They haven't updated |
| 17:23.32 | drmessano | Not IOS |
| 17:23.56 | drmessano | You keep arguing that this "broke with 10.3 and nothing works now" |
| 17:23.58 | drmessano | False |
| 17:24.23 | drmessano | Bria was updated, taking into account the 10.3 changes, and now backgrounds properly and does recieve calls |
| 17:24.40 | drmessano | I don't know why that triggers you so much |
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| 18:28.24 | jes__ | Hello does anyone have feedback on whether my expectation of asterisk is too high given the above scenario? I did a small test with a record and it was much worse. Down to 20-30 cps while save recordings to tmpfs mount. I just want to make sure there isn't something simple I'm overlooking. Would this be question better presented elsewhere? |
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| 18:55.40 | wabbits | jes__ what did you change to make it worse? |
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| 18:58.13 | wabbits | worse than what? |
| 19:11.15 | jes__ | I was just doing Answer, Playback (1 file), Moh, Hangup. I was getting 150 cps with sipp. I added a record so Answer, Playback (1 file), Record (5s), Moh, Hangup. With that scenario I get 20-30 cps. |
| 19:14.18 | wabbits | so the difference is Record (5s) |
| 19:14.49 | wabbits | try recording to /dev/null instead of to a file on tmpfs |
| 19:15.02 | wabbits | jes__ |
| 19:16.37 | jes__ | Okay, I'll give that a shot |
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| 19:43.57 | jes__ | Can you do that using the Record app? It certainly wants a file name with extension. I could create a /dev/null file and point it to that I suppose. |
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| 19:53.43 | jes__ | I made a file called new /dev/null file called it null.gsm and recorded to it. Seems to be getting the same results. |
| 19:54.02 | jes__ | I made a new /dev/null file called it null.gsm and recorded to it. Seems to be getting the same results. |
| 19:54.28 | jes__ | Though I am questioning that approach to recording to /dev/null |
| 19:57.41 | wabbits | jes__ Is the audio being transcoded from ulaw to linear16 then to gsm? |
| 20:05.13 | jes__ | Perhaps. I must admit I used gsm because it is the format of the audio files in /var/lib/asterisk/sounds/. I am not sure about the transcoding for Record(/filename.gsm) under the hood. I will add that to list of things to investigate for Asterisk. I'll rerun using ulaw format and see how it goes. |
| 20:15.28 | jes__ | Same. Now using "Record(/recording/${UNIQUEID}.ulaw,,8)". Sent in 30 cps for 4 min and started getting occasional timeout. Started getting timeouts after about 30s with a full call queue of about 500 calls. |
| 20:18.57 | jes__ | Stats at end were 6.5 min total 28.797 cps. 11,263 created calls. 484 failed. |
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| 20:32.15 | jes__ | So is this performance surprising to anyone or is it sound about right? I'm trying to gauge expectation of the software. |
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