| 00:04.56 | *** part/#asterisk kharwell (kharwell@nat/digium/x-ojtfgxarweoowwvb) |
| 00:08.07 | *** join/#asterisk Iamnacho (~Iamnacho@ip72-213-25-54.om.om.cox.net) |
| 00:47.24 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 01:19.56 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:19.56 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.5 (2017/12/22), Standard: 15.1.5 (2017/12/22); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 02:15.12 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 02:25.51 | *** join/#asterisk Dovid (~dovid@ool-45738ae3.dyn.optonline.net) |
| 02:31.20 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 03:35.37 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-zczfnynktdfabckc) |
| 03:43.42 | *** join/#asterisk simplydrew_ (~simplydre@unaffiliated/simplydrew) |
| 05:34.27 | *** join/#asterisk sh_smith (~sh_smith@cpe-76-174-26-91.socal.res.rr.com) |
| 05:54.13 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 06:02.23 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 07:22.11 | *** join/#asterisk aways (~aways___@unaffiliated/aways) |
| 07:22.15 | aways | Hi ! |
| 07:23.02 | aways | Question about template syntax. Have you a pointer on a doc ? What is the diff between template and include context directly ? |
| 07:41.19 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-xydigtdkymtydqfj) |
| 07:45.51 | *** join/#asterisk u0m3__ (~u0m3@86.127.131.167) |
| 07:49.59 | *** join/#asterisk u0m3_ (~u0m3@86.127.131.167) |
| 08:26.20 | *** join/#asterisk opmrcl (~Thunderbi@195.158.111.16) |
| 08:27.36 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 08:49.50 | *** join/#asterisk sekil (~sekil@cable-89-216-194-248.dynamic.sbb.rs) |
| 08:54.35 | *** join/#asterisk sekil (~sekil@cable-89-216-194-248.dynamic.sbb.rs) |
| 08:56.05 | *** join/#asterisk sekil (~sekil@cable-89-216-194-248.dynamic.sbb.rs) |
| 09:01.04 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 09:08.13 | *** join/#asterisk sekil (~sekil@cable-89-216-194-248.dynamic.sbb.rs) |
| 09:10.49 | *** join/#asterisk sekil (~sekil@cable-89-216-194-248.dynamic.sbb.rs) |
| 09:16.40 | *** join/#asterisk rwb (~Thunderbi@65.183.151.121) |
| 09:25.05 | *** join/#asterisk Mr_Pleb_Mgoo (~jakeb@103.46.213.148) |
| 09:31.59 | *** join/#asterisk detha (~detha@unaffiliated/detha) |
| 10:14.11 | *** join/#asterisk miralin (~Thunderbi@81.177.59.227) |
| 10:20.56 | *** part/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 10:30.03 | *** join/#asterisk opmrcl (~Thunderbi@195.158.111.16) |
| 10:40.29 | *** join/#asterisk opmrcl (~Thunderbi@195.158.111.16) |
| 10:42.01 | *** join/#asterisk Tim_Toady (~fuzzy@snf-33276.vm.okeanos.grnet.gr) |
| 10:43.33 | *** part/#asterisk opmrcl (~Thunderbi@195.158.111.16) |
| 10:46.22 | *** join/#asterisk dakudos (~dakudos@c-73-203-6-107.hsd1.co.comcast.net) |
| 11:34.04 | *** join/#asterisk miralin (~Thunderbi@81.177.59.227) |
| 11:53.57 | *** join/#asterisk jamesaxl (~James_Axl@109.172.62.242) |
| 11:56.34 | *** join/#asterisk lankanmon (~LKNnet@CPE64777dd7e053-CM64777dd7e050.cpe.net.cable.rogers.com) |
| 12:08.17 | *** join/#asterisk Worldexe (~Worldexe@95-107-33-134.dsl.orel.ru) |
| 12:32.01 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
| 12:36.50 | *** join/#asterisk polysics (~polysics@82.57.212.110) |
| 12:37.44 | polysics | hello everyone! still looking for a solution to tie media feeds to their users in Asterisk 15 SFU |
| 12:38.09 | polysics | I have been looking through the code but so far finding what registry to at least dump in the CLI has proven too hard for my limited knowledge |
| 12:38.24 | polysics | anyone that could help or we could hire to do so? |
| 12:40.52 | *** join/#asterisk BakaKuna (~BakaKuna@185.23.140.9) |
| 13:00.58 | *** join/#asterisk bitcoinastronaut (~monastery@2605:a601:ca4:2900:9c0:b3c6:a7fe:397d) |
| 13:19.59 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 13:41.37 | *** join/#asterisk jkroon (~jkroon@165.16.204.172) |
| 13:48.21 | *** join/#asterisk miralin (~Thunderbi@81.177.59.227) |
| 13:51.59 | *** join/#asterisk TheGallopingFox (TheGallopi@gateway/vpn/privateinternetaccess/thegallopingfox) |
| 13:52.23 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
| 14:05.27 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:08.19 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-qmdbevdfpqjwsnni) |
| 14:09.37 | *** join/#asterisk Zanelos (~zach@4.71.171.253) |
| 14:15.00 | *** join/#asterisk gtTuna (~Tuna_@d60-65-244-235.col.wideopenwest.com) |
| 14:20.42 | jamesaxl | hello |
| 14:21.20 | jamesaxl | Beackground sometimes does not work with some numbers, should I add Read also? |
| 14:21.38 | Samot | That is not a clear enough description. |
| 14:22.01 | Samot | What are the cases where it "sometimes doesn't work" vs "sometimes it works"? |
| 14:25.38 | jamesaxl | Samot: I have three incoming number 8872251-51-51, 88722555111 and 8963795-05-05, Background can give me choice when I click to a button with two 1st number, but with the 3rd the background continue without giving me any choice even I click button |
| 14:26.37 | Samot | OK, again not clear enough. |
| 14:28.26 | Samot | This is a dialplan issue. |
| 14:28.31 | *** join/#asterisk rwb (~Thunderbi@74.85.159.242) |
| 14:28.46 | Samot | So show the work dialplan for the two numbers that work and the dialplan for the non-working number. |
| 14:29.04 | jamesaxl | Samot: A moment please |
| 14:30.30 | file | you should also make sure that you are actually receiving DTMF from upstream... |
| 14:31.31 | Samot | ^^^ That too |
| 14:32.00 | jamesaxl | Samot: https://bpaste.net/show/23e23828b39e |
| 14:32.26 | *** join/#asterisk Dovid (~dovid@ool-45738ae3.dyn.optonline.net) |
| 14:32.27 | jamesaxl | file: I am going to check then |
| 14:32.49 | Samot | Are all three numbers from the same source? |
| 14:33.45 | jamesaxl | Samot: yep, I have Goto(05_services,s-Services,1) |
| 14:34.02 | Samot | No. I mean their provider. |
| 14:34.14 | Samot | This is going back to what file said about DTMF. |
| 14:34.32 | jamesaxl | Samot: providers are not same |
| 14:34.36 | Samot | Are all three numbers coming from the same provider and thus "trunk" |
| 14:34.40 | Samot | OK. |
| 14:34.58 | Samot | So it sounds like the trunk settings for this provider or this provider is having issues with DTMF |
| 14:35.08 | Samot | The DTMF mode is set to something they support? |
| 14:35.40 | jamesaxl | Samot: nop, I should check DTMF then |
| 14:37.22 | jamesaxl | Samot: numbers that do not work are gsm provider, I bought cards and I put the in the device. in the device every card is like SIP that I have to configure. |
| 14:38.58 | Samot | GSM? So does the provider expect you to be using a PBX with their service? |
| 14:40.24 | jamesaxl | Samot: of course it does |
| 14:41.12 | Samot | Not all GSM based services expect/accept inbound DTMF on the call. |
| 14:41.15 | Samot | GSM = mobile. |
| 14:41.57 | Samot | If I put my SIM card in a GSM device, it's still my service that I get on my phone. Which does not expect to handle/send inbound DTMF |
| 14:43.12 | jamesaxl | Samot: I have to contact provider then, and explain to them this issue |
| 14:47.13 | *** join/#asterisk waldo323 (~waldo323@75.151.31.89) |
| 14:55.44 | *** join/#asterisk jeffspeff (~Jeff@12.49.160.131) |
| 15:12.09 | *** join/#asterisk bounceman (~bounceman@185.32.9.250) |
| 15:12.45 | bounceman | Would it be possible to your knowledge to stop early media from upstream operator? In cases where the destination returns 404 our operator send back "The number you have dialed cannot be reached" with early media before sending a 404. |
| 15:13.06 | bounceman | So we are forced to listen to their message instead of handling it ourselves. |
| 15:19.22 | Samot | Is this a pure/straight Asterisk install? |
| 15:28.04 | bounceman | No, Kamailio are SBCs |
| 15:28.08 | bounceman | Asterisk in this case the B2BUA |
| 15:35.08 | Samot | Well I was checking if this was FreePBX or another distro. |
| 15:35.30 | Samot | Which would have that playback embedded for those failures to the caller. |
| 15:35.40 | Samot | But if you have Kamailio... |
| 15:35.57 | Samot | Then have Kamailio ACK that 404 and send your own to the system. |
| 15:37.01 | Samot | And that is interesting since most providers I've dealt with don't do that. |
| 15:37.28 | Samot | As in, insert early media like that. They just send the 404 and let you deal with it. |
| 15:38.02 | Samot | If you call that destination via your mobile/cell phone, do you get the exact same playback message? |
| 15:38.09 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-vklnrnxfpehtjuod) |
| 15:38.09 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 15:40.34 | jamesaxl | Samot: thank you for you help, even our provider allow pbx with gsm, DTMF from GSM to GSM is not allowed, I have to ask boos to buy trunks from pro sip provider |
| 15:40.40 | jamesaxl | s/boos/boss |
| 15:58.08 | bounceman | Samot: yeah, same message. Unfortunately our sip provider is the same as our mobile phone provider :-) |
| 15:58.11 | bounceman | So it makes sense |
| 15:58.21 | Samot | OK. |
| 15:58.28 | Samot | So call from another provider. |
| 15:58.42 | bounceman | price price price, I'll ask them if they can turn it off. |
| 15:58.44 | Samot | Verify the message is or is not the same. |
| 15:59.28 | Samot | If you're on ATT and get that message on all ATT services it could be ATT's message. |
| 15:59.57 | Samot | If you call from Sprint and get that exact same message, it could be the destination carrier sending that media back |
| 16:00.44 | Samot | However, if you call and the message is different in any way (down to the person recording the message) then yeah it's probably ATT sending that message back. |
| 16:00.48 | bounceman | Samot: it could also be that we do not bridge the legs before a 200 OK is sent. |
| 16:01.05 | bounceman | If we bridged the early media then it wouldn't be an issue since the message will be transmitted to the originating leg |
| 16:01.20 | bounceman | Now they have to wait 20 seconds in silent before a 404 is sent |
| 16:16.32 | *** join/#asterisk tzafrir (~tzafrir@bzq-84-109-19-7.red.bezeqint.net) |
| 16:17.36 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 16:39.55 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 16:58.13 | *** join/#asterisk salviadud (~ralfalfa@187-167-79-163.static.axtel.net) |
| 17:05.10 | *** join/#asterisk Worldexe (~Worldexe@95-107-33-134.dsl.orel.ru) |
| 17:12.12 | *** join/#asterisk LunaLovegood (~alice@75.98.139.193) |
| 17:12.39 | *** join/#asterisk roswell (roswell@roswell.systems) |
| 17:14.05 | *** part/#asterisk TheGallopingFox (TheGallopi@gateway/vpn/privateinternetaccess/thegallopingfox) |
| 17:14.06 | LunaLovegood | Sometimes I receive T.38 faxes from my ITSP (SIP trunk). Asterisk then complains about "Failed to initialize UDPTL, declining image stream". How can I make sure it's forwarded correctly to my ATA's? (G711 only, no T.38 support AFAIK). |
| 17:15.49 | roswell | LunaLovegood, what's udptl show config displays? |
| 17:18.01 | LunaLovegood | udptlstart: 4000 |
| 17:18.02 | LunaLovegood | udptlend: 4999 |
| 17:18.02 | LunaLovegood | udptlfecentries: 0 |
| 17:18.02 | LunaLovegood | udptlfecspan: 0 |
| 17:18.02 | LunaLovegood | use_even_ports: No |
| 17:18.04 | LunaLovegood | udptlchecksums: Yes |
| 17:18.23 | LunaLovegood | Oh dear, below 5000 is blocked in my firewall |
| 17:18.31 | roswell | looks good so far. and... |
| 17:18.40 | roswell | yup, was about to ask about ur fw settings |
| 17:18.50 | LunaLovegood | Can I set it to different values? |
| 17:19.10 | roswell | sure, in /etc/asterisk/udptl.conf |
| 17:20.42 | roswell | most likely you'll have to restart asterisk afterwards |
| 17:40.46 | *** join/#asterisk Zanelos (~zach@4.71.171.253) |
| 17:41.22 | Zanelos | Hi Asterisk people. I have a question. Can someone help me out with a conceptual issue I'm having with SIP? |
| 17:41.31 | roswell | just ask |
| 17:41.38 | Zanelos | Okie doke. |
| 17:42.27 | roswell | i mean, describe your issue instead of asking only abotu asking |
| 17:42.39 | Zanelos | So, I'm not understanding how an outbound proxy would work. The SIP RFC indicates that a proxy is never going to modify the To:/From: portion of the header, so how is it that you can have proxies like SER or Kamailio making routing decisions or performing LCR without modifying that portion of the header? |
| 17:43.55 | roswell | it's all in Via: header |
| 17:44.29 | Zanelos | roswell: Right. But at that point, Asterisk would be pointing to an upstream proxy as a trunk endpoint, and not a proxy. Right? |
| 17:45.17 | roswell | absolutely. asterisk doesn't distinguish between a proxy and a peer. to itself, these all are endpoints |
| 17:45.26 | roswell | or, peers |
| 17:46.03 | Samot | Well.. |
| 17:46.04 | Zanelos | Is that right? When I configured my upstream proxy, I specified a peer IP of my ITSP with the outboundproxy parameter pointing to my proxy. |
| 17:46.10 | Samot | One sec. |
| 17:46.11 | *** join/#asterisk dobson (~dobson@68.ip-149-56-14.net) |
| 17:46.17 | Samot | Let's break this down. |
| 17:46.49 | Samot | The TO header is the original destination sent by, in this case, a phone. |
| 17:47.27 | Samot | That holds the digits that were sent. Most Proxies/SBCs/etc do not allow that to be modified or manipulated in anyway. |
| 17:48.39 | Zanelos | Samot: I'm coming from a Cisco CUCM environment. The CUBE / CUCM will modify that portion of the header. For example, a CUCM will not accept calls that don't have it listed in the To: header. If the IP is wrong, it drops the message. |
| 17:48.42 | Samot | The FROM header holds the information about where the call came from. In the case of a SIP device it's generally the sip user and the domain. |
| 17:48.49 | Zanelos | So my understanding is kinda jacked up to begin with. |
| 17:48.54 | Samot | OK. |
| 17:49.08 | Samot | So if I call from my device which is 2112 to 2115 |
| 17:49.33 | Samot | The TO header will have 2115@pbx.domain.com |
| 17:49.59 | Samot | The FROM will be 2112@pbx.domain.com because these are on the same PBX... |
| 17:50.08 | Zanelos | Assuming that TN is registered to a device on the PBX, no. It only behaves that way for calls that go out over a trunk (Off-Net). |
| 17:50.14 | Samot | OK... |
| 17:50.22 | Samot | So now when I send that call to Asterisk... |
| 17:50.32 | Samot | There is what is called the Request URI header.. |
| 17:50.58 | Samot | That's what you see in the debug in the INVITE: sip: portion... |
| 17:51.29 | Samot | By default this is the Host set in the device. |
| 17:51.42 | Samot | If you set an Outbound Proxy in the device.. |
| 17:51.58 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 17:52.10 | Samot | The Request URI is change from 2115@pbx.domain.com to 2115@proxy.domain.com |
| 17:52.17 | Samot | So the call is sent there first. |
| 17:52.28 | Samot | The Proxy can look at the TO header.. |
| 17:52.48 | Samot | The proxy can look at the Request URI header or the FROM header and route on any of those items. |
| 17:53.05 | Samot | In the case of LCR, is re-writes that Request URI. |
| 17:53.38 | Samot | Or in the case of incoming routes from a Telco.. |
| 17:53.47 | Samot | You may have 23 numbers.. |
| 17:54.04 | Samot | Those numbers will always be in the TO header as dialed by the caller.. |
| 17:54.14 | Samot | But the Telco may send them all to one number. |
| 17:54.23 | Samot | So the Request URI is always the same number. |
| 17:54.51 | Samot | By default Asterisk looks at the Request URI aka the INVITE |
| 17:56.15 | Samot | When Asterisk sends a call to my Kamailio proxy and it does LCR routing... |
| 17:57.00 | Samot | The TO header is still DID@proxyip while the Request URI is DID@carrierip when it sends the call to the carrier. |
| 17:57.38 | *** join/#asterisk dadinck (~dadinck@216.1.237.162) |
| 17:57.44 | Samot | While the FROM has fromuser@asteriskdomain |
| 17:58.02 | Samot | So they know this call came from Asterisk, routes via the proxy to the carrier... |
| 17:58.43 | Samot | The carrier knows the original source while know it has to send calls to the proxy to get there. |
| 17:58.53 | Samot | Aka the Record Route |
| 18:10.31 | Zanelos | Samot: thanks for the info. I'll digest and probably follow up with you |
| 18:19.12 | *** join/#asterisk miralin (~Thunderbi@81.177.59.227) |
| 18:29.25 | *** join/#asterisk miralin (~Thunderbi@81.177.59.227) |
| 18:33.04 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 18:47.17 | *** join/#asterisk robink (~quassel@unaffilated/robink) |
| 19:07.25 | Zanelos | Samot: So, if that's the case, what info did you provide the ITSP? |
| 19:07.56 | Zanelos | That's a big chunk of my confusion. When you tell an ITSP that signalling will come from a certain IP, would it be the proxy IP, or the PBX's IP? |
| 19:08.21 | Zanelos | And thank you by the way. This has been hugely instructional. |
| 19:17.27 | Zanelos | err instructive* |
| 19:45.53 | *** join/#asterisk salviadud (~ralfalfa@187-167-79-163.static.axtel.net) |
| 19:52.50 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-kmnfglksqfeptzay) |
| 19:52.50 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 21:42.36 | *** join/#asterisk rwb (~Thunderbi@65.183.151.121) |
| 22:23.09 | *** part/#asterisk dadinck (~dadinck@216.1.237.162) |
| 22:31.15 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:02.15 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 23:32.21 | *** join/#asterisk znoteer_ (~quassel@157.52.8.172) |
| 23:40.41 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-nzgpjrctsgxrcrdx) |
| 23:58.25 | *** join/#asterisk lankanmon (~LKNnet@99.245.204.29) |