| 13:53.41 | *** join/#asterisk infobot (ibot@rikers.org) |
| 13:53.41 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.9.1 (2016/05/13), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 13:55.33 | tcpip6 | I this have the: SIP_CODEC=g729 option to explore... not sure there is a equivalent for IAX channels |
| 13:57.14 | [TK]D-Fender | none that I'm aware of |
| 13:57.45 | tcpip6 | BTW, going pure g729 without the codec (only format_g729) is working well the everything.. voicemail included... the only issue that remains for me is testing with originate of testsing with a spool call file |
| 13:58.17 | tcpip6 | but then again... this is not a show stopper prob |
| 13:58.41 | *** join/#asterisk jeffspeff (~Jeff@64.233.156.242) |
| 13:59.04 | [TK]D-Fender | Maybe for the things you need... but as asoon as you need anything transcoded or generated (like inband indications) then you're up a creek |
| 14:00.51 | igcewieling | tcpip6: asterisk transcods a LOT, dialtone, ringing tone, IVR, busies, etc. |
| 14:01.49 | igcewieling | We used to have all sorts of issues with transcoding to g729 when we didn't want to, but the fix was easy. |
| 14:02.24 | igcewieling | The fix is to buy some damn G729 licenses (or in our case a couple of transcoding cards with g729) |
| 14:03.46 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 14:03.52 | tcpip6 | In may case, I am using Sipura PAP2T boxes with a small ARM based (OpenWRT) router to connect to a SIP termination provider.. and I use IAX for calls between users... |
| 14:04.22 | tcpip6 | works fine.. I have to set the silence detection in the voicemail app to 0 |
| 14:04.26 | [TK]D-Fender | "between users"? |
| 14:04.38 | [TK]D-Fender | who are "users" vs those PAP2T's? |
| 14:04.51 | tcpip6 | and I made use that all the sound files are in native g729 |
| 14:05.51 | tcpip6 | friends and family.. each install only has 1 PAP2T (FXS box) |
| 14:08.37 | *** join/#asterisk kharwell (kharwell@nat/digium/x-fqgmzyowkfpvhhmc) |
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| 14:21.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:22.55 | jeffspeff | I have a fax machine (fax/printer/scanner/copier) connected to a cisco ata device. i have disabled fax detection on the sip peer of the ata, disabled it on my providers peer and disabled it in my general sip.conf settings. when i try to receive a fax this is what i get. http://pastebin.com/pUq2Lp5J |
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| 14:26.42 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:29.15 | *** join/#asterisk overyander (~overyande@12.49.160.131) |
| 14:30.55 | [TK]D-Fender | Doesn't mean THEY didn't try offering T.38 |
| 14:31.09 | [TK]D-Fender | And you're bringing us this... WITHOUT SIP debug..... |
| 14:35.48 | jeffspeff | just a min |
| 14:38.28 | tcpip6 | Small update... using ",Set(SIP_CODEC=g729)" |
| 14:38.55 | tcpip6 | the system still tries to use SLIN .. |
| 14:41.04 | tcpip6 | so I guess any kind of call origination is not gonna happen without trancoding |
| 14:41.17 | tcpip6 | oh well... |
| 14:42.53 | *** join/#asterisk hdon (~hdon@wsip-24-234-30-238.lv.lv.cox.net) |
| 14:43.35 | hdon | hi all :) can anyone recommend a softphone for windows that allows multiple simultaneous extensions and calls and can associate a given extension to a given audio device? |
| 14:44.04 | hdon | we're using x-lite and i thought i might be able to run multiple instances of it, but it didn't quite work out that way |
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| 14:51.25 | SpaceInvaders | Hello! In the Fedora 23 package for Asterisk the only deps that come up (dnf install asterisk) are ilbc and pjproject. Are the deps listed in The Definitive Guide no longer required because compiling isn't required? |
| 14:51.42 | igcewieling | I don't know about seperate audio devices, but I believe the commercial version of x-lite might do what you want. |
| 14:53.39 | jeffspeff | is there a way to get the debug log for a specific peer to be logged in a specific log file? |
| 14:58.11 | [TK]D-Fender | jeffspeff, No |
| 14:58.19 | [TK]D-Fender | not from * |
| 14:59.02 | [TK]D-Fender | SpaceInvaders, Or because the packages weren't built with them so you are doing without some functionality |
| 15:01.28 | SpaceInvaders | Is there any documentation on the Asterisk packaging by distro? |
| 15:02.09 | tcpip6 | Ok I am off.. thanks for the help [TK]D-Fender |
| 15:02.12 | *** part/#asterisk tcpip6 (4637bbf1@gateway/web/freenode/ip.70.55.187.241) |
| 15:03.19 | [TK]D-Fender | Depends on what the packagers chose to do. TThat isn't "us" |
| 15:03.41 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 15:04.25 | SpaceInvaders | Got it--I'll check with #fedora. Does my obi110 qualify as a "physical PSTN interface" requiring DAHDI and LibPRI? I'm guessing the answer is "yes" but just checking to be sure. |
| 15:05.48 | [TK]D-Fender | no |
| 15:06.02 | SpaceInvaders | Thanks for your help! |
| 15:06.04 | [TK]D-Fender | those are for DAHDi compatible PCI cards |
| 15:06.23 | SpaceInvaders | ok I've seen those, before. |
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| 15:23.28 | st2 | hello. i use in my extension dial command like that: Dial("SIP/sip_new-pops_1-00001cc7", "SIP/new-pops_102&SIP/new-pops_101&SIP/new-pops_103,60,g|M(logger)|t") can i findout in macro which channel was available during the call? |
| 15:25.04 | [TK]D-Fender | <PROTECTED> |
| 15:26.41 | st2 | for example i see logs like that: |
| 15:26.42 | st2 | [2016-06-21 18:13:51] WARNING[15773][C-000022db]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
| 15:26.43 | st2 | <PROTECTED> |
| 15:26.43 | st2 | <PROTECTED> |
| 15:26.43 | st2 | <PROTECTED> |
| 15:27.00 | *** join/#asterisk mirela666 (~mirkob@89.184.168.160) |
| 15:27.17 | [TK]D-Fender | Check before you dial then |
| 15:27.22 | st2 | two subscriber was unavailable during the call. number 102 was in ringing state |
| 15:27.30 | [TK]D-Fender | And build your dial up based on that. |
| 15:28.39 | jeffspeff | [TK]D-Fender as requested, here's the sip debug for that peer. http://pastebin.com/yANSYPC8 it has been sanitized, let me know if you would like unsanitized and i will pm you another link |
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| 15:29.19 | st2 | so how i can check it in extension? |
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| 15:31.32 | [TK]D-Fender | st2, "core show function DEVICE_STATE" |
| 15:31.48 | [TK]D-Fender | jeffspeff, I'm not sure on this, but hopefully someone else can chime in |
| 15:32.07 | st2 | thanks! |
| 15:32.49 | jeffspeff | I have a fax machine (fax/printer/scanner/copier) connected to a cisco ata device. i have disabled fax detection on the sip peer of the ata, disabled it on my providers peer and disabled it in my general sip.conf settings. when i try to receive a fax this is what i get. http://pastebin.com/yANSYPC8 |
| 15:33.37 | st2 | but is it possible to know afterwards of call in macro which channel was in ringing state? |
| 15:35.34 | [TK]D-Fender | st2, very complicated, but possible if you dial them all as local channels and have the inside dial's have no ttimeout and check the dialstatus if it bombs immediately or not and set an inherited channel var |
| 15:35.47 | [TK]D-Fender | very messy, but should be doable |
| 15:38.59 | st2 | thanks! i think first solution is more correct |
| 15:39.05 | st2 | in my case |
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| 15:59.30 | subvhome | hello all.. I'm new to asterisk and was wondering what is needed to get started. I have Asterisk installed but not sure what else needs to be done. I want 4 lines in my home to be able to dial out to the world and the world to dial in to my lines.. |
| 16:00.17 | subvhome | I suppose I need a SIP gateway correct? SIP Phones/Soft Phones, a server running asterisk... and a tutorial to set it all up |
| 16:05.21 | TandyUK | subvhome: or an FXO card to put in your asterisk box |
| 16:05.45 | TandyUK | or a combined FXO (Line) and FXS (handset) card |
| 16:06.24 | TandyUK | analog telephony is so last century though, i'd look at getting a voip trunk, and some phones (whether actual voip phones, or soft phones) |
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| 16:08.33 | [TK]D-Fender | first be clear about how many TELCO lines you are talking about (links to the phone company), vs phones,etc |
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| 16:13.18 | subvhome | well I currently have 1 number... |
| 16:13.30 | subvhome | was hoping to set extensions in my home |
| 16:14.09 | ||cw | you can do that, but then you still only get 1 call at a time |
| 16:14.52 | subvhome | I guess for my trial run.. i will work with that.. and then figure out what i need to do in order to get more lines.. |
| 16:15.48 | subvhome | since its my first time with this.. i decided the use asteriskNOW... should be enough to get me going correct? |
| 16:15.57 | [TK]D-Fender | Your phones are another question. |
| 16:16.18 | [TK]D-Fender | You could have them all hooked up to ONE port on an FXS interface of some sort and share it like they do on your home line now |
| 16:16.48 | [TK]D-Fender | or you could redo your wiring to allow each to connect to a separate port on independent FXS ports |
| 16:17.06 | ||cw | or switch to all IP phones |
| 16:17.10 | [TK]D-Fender | Ditch AsteriskNOW and get the official FreePBX ISO from their site |
| 16:17.17 | TandyUK | does a single FXS port have enough power to give REN4? |
| 16:17.25 | ||cw | or get a multiline cordless system |
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| 16:17.45 | subvhome | I will start with freepbx iso |
| 16:18.02 | ||cw | TandyUK: depends on the device. |
| 16:18.08 | TandyUK | if youre a newbie, id recoomend something with a nice gui :P |
| 16:18.27 | TandyUK | asterisk config files arent the easiest things to work with :P |
| 16:18.32 | ||cw | TandyUK: isn't that what freepbx is? |
| 16:18.39 | TandyUK | no idea, ive never used it |
| 16:20.15 | [TK]D-Fender | Any decent device should handle that REN |
| 16:20.33 | [TK]D-Fender | IIRC most of the popular ATA's have an REN of about 7 |
| 16:22.19 | TandyUK | oh cool |
| 16:22.38 | igcewieling | analog (n): 1) what you will be disappointed with if used with Asterisk. |
| 16:22.38 | TandyUK | any recommendations on a super simple ata for a fax machine |
| 16:23.22 | TandyUK | upgrading a client from an ISDN based system to voip, and they currently have a couple of fax machines theyre not goign to replace |
| 16:24.01 | TandyUK | most ive convinced them to bring into the 21st century with email-to-fax and fax-to-email :P |
| 16:24.12 | TandyUK | but these 2 they are adamant they cant replace |
| 16:24.58 | ||cw | TandyUK: does the voip do t.38 properly? |
| 16:25.41 | igcewieling | you will want analog lines for those faxes |
| 16:25.42 | TandyUK | we also have a couple of door entry phones, which currently connect to analog ports on their avaya |
| 16:25.58 | ||cw | fax is a bitch on voip |
| 16:26.25 | TandyUK | fax is just a bitch |
| 16:26.35 | igcewieling | seems to work just fine on analog |
| 16:26.37 | TandyUK | i dont understand why people have to fax stuff to people they email already |
| 16:26.55 | ||cw | t38 makes it better, g711 on a low latency/low hop count can work OK if you force the baud slower. anything else is just not gonna work out |
| 16:27.15 | ||cw | TandyUK: it's easier than scanning to email |
| 16:27.34 | ||cw | also some fools don't realize that you can |
| 16:27.36 | TandyUK | when its a document they print off first lol |
| 16:27.46 | TandyUK | just attach pdf to email and be done |
| 16:27.59 | TandyUK | currently they print it, then fax it to recipient, who no doubt then scans it in again |
| 16:28.13 | ||cw | we had an issue with a fax server once (modem failure), and one customer was like "oh I can email it? I'll just do that instead" |
| 16:28.13 | TandyUK | just seems like a waste of time and paper to me :P |
| 16:28.29 | igcewieling | I find it easier to not tell the customer they are doing it wrong and to spend lots of money in hardware and training just to send faxes. |
| 16:28.41 | ||cw | TandyUK: that doens't work if you have to sign it |
| 16:28.50 | TandyUK | echosign :P |
| 16:29.16 | TandyUK | theres literally no need for faxes in the 21st century imho |
| 16:29.43 | TandyUK | even the laws in the uk at least have changed enough to make email and fax no different |
| 16:29.46 | ||cw | that one little outage is what it took to get most of our faxes to email. we're down to about 10 faxes a day, from a peak of over 100 |
| 16:30.06 | igcewieling | TandyUK: Of course there is a need. Customers want them. |
| 16:30.23 | TandyUK | no they dont, they just dont realise theres a better way |
| 16:31.01 | ||cw | they don't want to learn something new |
| 16:31.39 | igcewieling | They don't want to learn something new and I don't want to support it. With a fax machine I don't have to support it. |
| 16:31.57 | ||cw | what helped here is using a network scanner that kind of looks like a fax machine. |
| 16:33.17 | igcewieling | anyway, I'm off to relocate a squirrel, be back in a bit. |
| 16:33.41 | WIMPy | It's quite shocking how many people seem to be unable to configure their PBX to accept faxes for their fax machine. |
| 16:35.26 | WIMPy | (or send them) |
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| 16:37.47 | ||cw | WIMPy: it's usually the sending to the other end that's the real issue, and it's often hard to tell |
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| 17:14.05 | _boot | hi all, i'm trying to connect an sip softphone (baresip) to asterisk with tls, it's registering fine but it won't make any calls - i'm given a 488 Not Acceptable Here. from the sip debug log it doesn't even look like the other end is contacted... Is there something obvious I'm missing? |
| 17:16.51 | _boot | One thing I've noticed is that the INVITE is to sip:1001@192.168.0.17;transport=tls, but 1001 isn't using TLS. Could this be the issue? (also, would that be a client bug or something?) |
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| 17:34.38 | [TK]D-Fender | if you're trying and it says it isn't using it... it isn't going to work |
| 17:42.53 | _boot | the phone i'm calling from is using TLS, though - that's what I'm trying to achieve |
| 17:47.07 | [TK]D-Fender | Then configure your peer tto match |
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| 17:58.48 | _boot | I have - it's the remote end of the call that isn't using tls, but that shouldn't matter right? Even so, the sip logs don't appear to be contacting the remote end at all? http://pastebin.com/fL4cpmvC |
| 18:00.19 | igcewieling | have you made sure all devices trust the certificate? |
| 18:01.01 | igcewieling | _boot: sip debug is useless for determining which peer was matches. |
| 18:01.05 | _boot | the caller is the only thing using tls, (ext. 2001) - if it didn't wouldn't it be unable to initiate a call? |
| 18:01.30 | _boot | okay, should I turn all debug on and try again? |
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| 18:04.16 | jeffspeff | I have a fax machine (fax/printer/scanner/copier) connected to a cisco ata device. i have disabled fax detection on the sip peer of the ata, disabled it on my providers peer and disabled it in my general sip.conf settings. when i try to receive a fax this is what i get. http://pastebin.com/yANSYPC8 |
| 18:29.12 | _boot | aha! So in the debug logs (after a message about sending a 488?) there's this little guy: handle_request_invite: No compatible codecs for this SIP call. |
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| 18:34.10 | _boot | hmm, i'm going to try without using tls |
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| 19:52.13 | jeffspeff | in order for an ata device to handle its own t.38 fax calls, do i need to enable t.38 as a codec or something? i want * to stay out of the way and just pass the call into the ata |
| 19:59.11 | ||cw | jeffspeff: you've seen https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway right? |
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| 20:04.34 | jeffspeff | ||cw thanks, let me try that |
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| 20:17.16 | igcewieling | jeffspeff: it is my understanding, re-inviting away from Asterisk do the device talks directly to the carrier is not expected to work with T.38 |
| 20:17.40 | igcewieling | s/do/so/ |
| 20:42.08 | _boot | hmm so it seems my issue currently is asterisk not finding any compatible codecs for a call. the phone i'm using is listing g722,pcmu,pcma and gsm in sdp. on a successful debug log (on another device) i can definitely see at least gsm in the "Capabilities - us" list after sdp parsing. can anyone suggest what on earth might be going on? |
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| 21:41.08 | overyander | igcewieling, any ideas what happened here? the cli didn't show any errors. this is from the packet capture of a fax attempt. http://pastebin.com/434WhKxr |
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| 21:45.55 | overyander | that's the result now that i'm using the faxopt gateway option. it's accepting the t38 connection, negotiates a modem and then just says bye |
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| 22:12.22 | SpaceInvaders | Should I stick with ALAW if I'm in the US but frequently make and receive calls to EU? |
| 22:12.48 | SpaceInvaders | or should I install ULAW as The Definitive Guide states? |
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| 22:19.58 | rmudgett | There is no harm in installing both. Those formats are tiny in comparison to other codecs. To translate between alaw and ulaw is just a table lookup. |
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| 22:24.41 | SpaceInvaders | Thanks! |
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| 23:22.59 | igcewieling | overyander: there are many things I will do, but helping with fax is not one of them. |
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| 23:35.30 | overyander | igcewieling lol, thanks anyways |
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