| 00:08.45 | ruied | has anyone worked with zyxel's 72port sip to analog card "VOP1372G-61" (http://www.zyxel.com/products_services/ies_6000_series.shtml?t=p) |
| 00:12.47 | ruied | I've worked with the zyxel's adsl2 cards and they are working for several years without a single problem. I wonder if the SIP cards area also fine and if they are ok for asterisk.... |
| 00:24.15 | *** join/#asterisk dijib (~dijib@24-138-184-76.eastlink.ca) |
| 00:24.33 | dijib | SeRi: are you in here? |
| 00:24.39 | SeRi | yes |
| 00:24.44 | dijib | sweet |
| 00:24.59 | dijib | someone from a 24.x.x.x address was trying to access asterisk. maybe penguine? |
| 00:25.03 | dijib | im in conf fyi |
| 00:25.08 | SeRi | I saw that one |
| 00:25.33 | dijib | hey see the other |
| 00:25.36 | dijib | ? |
| 00:25.40 | dijib | minexpire error |
| 00:25.41 | SeRi | one sec |
| 00:26.00 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 00:26.01 | SeRi | I saw it |
| 00:26.06 | SeRi | all of it as it was happening |
| 00:26.15 | SeRi | why would it be Penguin? |
| 00:26.20 | dijib | i switched to csipsimple today from sipdroid. maybe that? |
| 00:26.26 | dijib | the 24. might be i dont know |
| 00:26.31 | dijib | doesnt make sense really |
| 00:26.43 | SeRi | chan_sip.c:25120 handle_request_subscribe: Received subscription for extension "asterisk" context "phones" with Expire header less that 'minexpire' limit. Received "Expire: 30" min is 60 |
| 00:26.55 | dijib | ya |
| 00:26.59 | SeRi | is that your peer? |
| 00:27.09 | SeRi | for your cell |
| 00:27.13 | dijib | nope |
| 00:27.20 | dijib | 400 is peer |
| 00:27.21 | SeRi | so than somebody was hitting you |
| 00:27.32 | dijib | imma hit somebody |
| 00:27.43 | SeRi | look at your cdr |
| 00:35.11 | *** join/#asterisk JimDickenson (~dickenson@c-71-56-148-165.hsd1.wa.comcast.net) |
| 00:40.33 | dijib | SeRi: no calls made it into the CDR. |
| 00:40.42 | dijib | which means they did not reach my outbound context |
| 00:53.59 | *** join/#asterisk greenwolf (186788ee@gateway/web/freenode/ip.24.103.136.238) |
| 00:54.17 | cheasee | im playing with chan_dongle, actualy trying to get it run under freepbx working virtualized with qemu-kvm. when initializing the dongle, i get this error: ERROR[22294]: chan_dongle.c:433 do_monitor_phone: [privatone] timedout while waiting 'OK' in response to 'AT' |
| 00:55.08 | cheasee | (same setup is working on a real host and same setup virtualized with vmware esxi5 works too) |
| 00:55.25 | cheasee | anybody ideas what/where i could debug further? |
| 00:56.17 | greenwolf | its probably a problem with qemu emulator |
| 00:57.59 | ChannelZ | Is that Hayes? |
| 01:00.52 | cheasee | hayes? |
| 01:00.54 | WIMPy | You don;t seem to have access to the USB port, I guess. |
| 01:01.23 | WIMPy | yes |
| 01:01.42 | WIMPy | Normal modem style. |
| 01:01.49 | dijib | dongle for what? |
| 01:02.15 | WIMPy | gsm |
| 01:02.19 | cheasee | im able to access the data device via screen |
| 01:02.27 | cheasee | uh, not modem, gsm |
| 01:02.45 | cheasee | huawei e1550 usb dongle |
| 01:03.13 | WIMPy | "modem" |
| 01:03.52 | WIMPy | DCE |
| 01:04.06 | ChannelZ | +++ATH |
| 01:05.25 | cheasee | ah right, true |
| 01:07.11 | gusto | so |
| 01:07.18 | gusto | who has a e1550 here? |
| 01:07.22 | gusto | i have such thing as well |
| 01:07.31 | gusto | and here i have an E220 |
| 01:08.10 | cheasee | ive got an e1550 and an e180 working :) |
| 01:08.16 | WIMPy | E172 and E1550 |
| 01:08.34 | gusto | E173 i think it is what i have as well |
| 01:08.36 | cheasee | e122 doesnt work, tried everything, even firmware update |
| 01:08.37 | gusto | not E172 |
| 01:09.29 | gusto | but however, these devices are really cool, the only problem about them is the poor provider's network |
| 01:10.01 | WIMPy | My e1550 won't do voice, either. |
| 01:10.10 | gusto | i mean with t-mobile and vodafone you at least get good signal, but then it's all NATted and proxyservered, that's no good internet connection |
| 01:10.23 | gusto | i do not think that any of mine does |
| 01:10.36 | gusto | but that E220 is old, so theoretically that would be the best shot to try |
| 01:10.52 | cheasee | "three"/3 gives me a public ip :) |
| 01:10.52 | dijib | i hope you have a 25ft cable and an enclosure for that e1550 |
| 01:11.01 | dijib | 18 |
| 01:11.08 | gusto | the older the device the less likely is it antifeautured |
| 01:11.22 | gusto | cheasee: in austria? |
| 01:12.50 | dijib | gusto its registering in dmesg properly? getting irq? |
| 01:13.38 | cheasee | dijib: 25ft cable? enclosure? |
| 01:13.44 | gusto | what? |
| 01:13.45 | cheasee | gusto: yes, in austria |
| 01:13.53 | dijib | so you can put that thing outside. |
| 01:13.55 | gusto | cheasee: that's even worse |
| 01:14.13 | gusto | from what i heard austria has no good wireless networks |
| 01:14.32 | gusto | i could have HSPA+ with 42 MBit/s here! ;-) |
| 01:14.43 | gusto | but i do not have a dongle that could do that |
| 01:14.44 | cheasee | dijib: ive got enough signal to get calls through without problems, so no must need to put it outside |
| 01:14.58 | dijib | find one in the garbage |
| 01:15.48 | cheasee | gusto: austria has no good wireless networks? |
| 01:15.53 | gusto | cheasee: yes |
| 01:16.00 | cheasee | define please ;) |
| 01:16.21 | gusto | well, they are old and outdated, like everything there |
| 01:17.47 | cheasee | gusto: where are you from? is it that much better where you live? |
| 01:18.08 | gusto | slovakia has much better UMTS coverage |
| 01:18.31 | gusto | you see it on my domain where i am from now |
| 01:18.36 | gusto | t-com.sk |
| 01:18.54 | cheasee | i dont whois everybody im speaking to ;) |
| 01:19.34 | cheasee | well isps provide lte lately.. ? |
| 01:20.10 | gusto | http://www.telekom.sk/english/company/history/ |
| 01:20.11 | cheasee | a friend got hspa+ with 25mbit (outside cities where less people use 3g) |
| 01:20.39 | gusto | http://www.dsl.sk/images/articles/2011-10-31-orange-dc-1.jpg |
| 01:21.49 | dijib | this: BBi5 -> [Jan 5 20:16:07] WARNING[10128]: chan_sip.c:25120 handle_request_subscribe: Received subscription for extension "asterisk" context "phones" with Expire header less that 'minexpire' limit. Received "Expire: 30" min is 60 |
| 01:22.06 | gusto | http://www.o2.sk/pre-vas/internet/lte |
| 01:22.30 | dijib | while sending an SMS+image apparently being caught by asterisk from the cellphone. |
| 01:22.34 | gusto | that's LTE but only in few locations ... not slovakia-wide operation |
| 01:23.55 | cheasee | gusto: http://www.a1.net/hilfe-support/netzabdeckung/ |
| 01:24.06 | cheasee | type in "wien" and search |
| 01:24.52 | cheasee | it displays whole austria.. im not sure, but on first sight it looks better than in .sk ? |
| 01:25.49 | gusto | cheasee: that's A1 and that's vodafone |
| 01:26.39 | gusto | but that map you gave me is not realistic, because you can get those speeds only for some distance of the BTS |
| 01:27.26 | gusto | that orange network i gave you is 1.) realistic 2.) shows ONLY HSPA+ ... you have ALL UMTS there |
| 01:27.43 | gusto | is zu 42 Mbit/s <-- that can mean even 7,5 mbit/s |
| 01:28.02 | gusto | bis zu 236 Kbit/s <-- that looks like GSM !!! |
| 01:28.11 | cheasee | ok |
| 01:28.30 | gusto | you understand what i mean? |
| 01:28.52 | cheasee | i know what "up-to" means ;) |
| 01:28.56 | gusto | yes |
| 01:29.07 | cheasee | i worked for an isp and had to explain that to many customers :P |
| 01:29.16 | gusto | but you have bis zu 236 Kbit/s and that is not even UMTS, so it's GSM with some extensions |
| 01:29.49 | gusto | and bis zu 42 Mbit/s can mean UMTS basic 386kbps to HSPA+ and HSPA+ DC ! |
| 01:30.12 | cheasee | sure :) |
| 01:30.26 | gusto | so on first sight it looks better than that orange.sk, but that orange.sk map is a map for HSPA+ ONLY! |
| 01:30.39 | cheasee | and you always get those ~40mbit constantly? |
| 01:31.05 | gusto | http://www.orange.sk/private/coverage/index.jsp?locale=sk |
| 01:31.07 | gusto | here! |
| 01:31.10 | gusto | take that |
| 01:31.11 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 01:31.33 | cheasee | i dont have much experience with all networks, just with orange, t-mob and three.. three is the fastest one |
| 01:31.46 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 01:32.13 | gusto | when you choose GSM/GPRS/EDGE and "zobraz" then austria does not look soo good any more :-D |
| 01:32.21 | cheasee | i dont like orange at all heh, three is nice, it even lets you phone for free within their network :) |
| 01:32.35 | cheasee | zobraz ? |
| 01:32.40 | gusto | you can not compare orange in austria and orange in slovakia |
| 01:32.49 | gusto | that were always two very different companies |
| 01:33.05 | gusto | orange in austria (which does not exist any more) was using old Siemens hardware |
| 01:33.38 | cheasee | aha? interesting |
| 01:33.57 | cheasee | yes, three/hutchison bought orange in austria |
| 01:33.58 | gusto | and here Orange has all Alcatel-Lucent and Nortel hardware and planning was done by some dudes from ESA |
| 01:34.31 | gusto | but now they ll be using Huwaei like everyone else |
| 01:35.25 | gusto | yes, then 3 will need to exchange all BTSs from orange there |
| 01:37.57 | gusto | so wireless networks are no problem here in slovakia, but DSL is ... having the german telecom here as monopolistic provider for all telephone and DSL lines is worse than russian occupation, because they were not staying forever |
| 01:38.18 | dijib | Penguin: your alive, i didnt see you entres |
| 01:40.03 | cheasee | hah, here in austria dsl is no problem, telecom even gives you 7km dsl line with "up-to 16/1mbit" hahaha and you end up having 1-2mbit down when the cabling isnt rusty haha |
| 01:40.07 | dijib | also did anybody see my sms issue? is this possible through itsp's? sms? |
| 01:40.39 | Maliuta | dijib: it depends on the itsp ... mine has an API for doing it |
| 01:41.41 | gusto | cheasee: in slovakia you get DSL everywhere as well, .... for 40 EUR / month !!! LOL :-D |
| 01:42.12 | cheasee | omg? for 40eur you get cable/vdsl in austria |
| 01:42.26 | gusto | you see |
| 01:42.46 | gusto | that's deutsche telekom thiefs and swindlers inc. |
| 01:43.19 | cheasee | heh |
| 01:43.39 | gusto | i had DT 2 mbit/s in nuernberg as well, so they are not better out there, now i have 18/1 MNet there |
| 01:43.47 | gusto | with native IPv6, what i miss here |
| 01:44.35 | file | I'm 80Mbps down, 30Mbps FTTH here... wish it had native IPv6 but I settle for an he.net tunnel |
| 01:44.46 | gusto | yes yes |
| 01:44.55 | file | thankfully my ISP peers with them so the extra latency is minimal |
| 01:45.05 | gusto | is possible here in this city as well, but not to my address |
| 01:45.07 | cheasee | ya, native v6 in austria only possible with sil.at i couldnt find another isp with native v6 support |
| 01:45.07 | file | eerily Google over IPv6 via the tunnel has less latency than IPv4 directly |
| 01:45.30 | gusto | http://www.orange.sk/web/internet/internet_na_doma/fibernet-aktualna-ponuka.html |
| 01:45.40 | gusto | that's 100/100 for 30 EUR |
| 01:46.16 | gusto | no one can beat that |
| 01:46.28 | cheasee | file: lol! |
| 01:47.07 | file | 40ms over IPv6, 50ms over IPv4 |
| 01:47.12 | *** join/#asterisk engrxyz (~rewra@host81-150-217-167.in-addr.btopenworld.com) |
| 01:47.43 | gusto | these *** get 100/100 for 10 eur less than me for my t-com 10/0.5 DSL only few km away, and maybe less than one by air |
| 01:48.02 | gusto | file: i have such data as well |
| 01:48.04 | file | a few km is a lot of fiber ^_^ |
| 01:48.10 | cheasee | gusto: i guess thats only possible in new buildings where fiber is already available? or does orange put fiber in old buildings? |
| 01:48.15 | gusto | 64 bytes from fa-in-f138.1e100.net (173.194.70.138): icmp_seq=1 ttl=48 time=39.7 ms |
| 01:48.25 | gusto | 64 bytes from fa-in-x65.1e100.net: icmp_seq=1 ttl=57 time=51.4 ms |
| 01:48.43 | gusto | 64 bytes from 2a00:1450:4001:c02::65: icmp_seq=1 ttl=57 time=50.6 ms |
| 01:48.58 | gusto | cheasee: yes |
| 01:51.44 | gusto | cheasee: http://dopice.sk/4GK |
| 01:53.19 | cheasee | interesting, such buildings here in austria are the last who get fiber .. |
| 01:54.02 | gusto | here it only depends on how many ppl will buy it |
| 01:54.36 | gusto | and buildings with high density of potential customers are likely to be in the first row |
| 01:55.07 | cheasee | that would be logical, but somehow thats not the case in austria (or at least in vienna) |
| 01:55.36 | gusto | in vienna noone would care about tearing off the street for putting cables in there |
| 01:56.56 | cheasee | there are much more potential customers where fiber has to be passed but it costs too much to pass "so much" fiber.. its a war with local regulators :/ |
| 01:57.43 | cheasee | true :/ but there are comapnies like cablerunner who use alternatives heh |
| 02:00.19 | cheasee | gusto: http://www.cablerunner.com/technologie/rohrkanaele.html |
| 02:04.23 | gusto | that's not an alternative to run the fibre optics in the shitter |
| 02:06.21 | WIMPy | This analog shit seems to like me just as much as I like it. |
| 02:06.24 | gusto | but that's typical for the austrian ppl, they invent bullshit because they are too lazy |
| 02:07.08 | gusto | but who else would get that idea to pass fibreoptics in canalisation? |
| 02:07.26 | WIMPy | That's not a new idea. |
| 02:07.35 | WIMPy | The nerds in Berlin have been doing so for decades. |
| 02:07.39 | gusto | aha |
| 02:07.44 | gusto | another lazy ppl |
| 02:07.47 | gusto | berlin! |
| 02:07.49 | WIMPy | Well not with fibers at that time, obviousely. |
| 02:09.31 | WIMPy | Here our utility company put fibers in to the old gas pipes. |
| 02:09.46 | cheasee | gusto: lol? you prefer a loud building lot over weeks than this idea? |
| 02:10.01 | gusto | well, thats understandable, i would do the same, but those gas pipes are OK |
| 02:10.18 | gusto | of course i would prefer that |
| 02:10.38 | gusto | would do that by myself preferably |
| 02:11.06 | gusto | pneumatic hammer is fun |
| 02:11.56 | gusto | once we built a cluster of two compressors to tear up some concrete and while we were talking we forgot about the pressure and one of them blew up |
| 02:12.29 | gusto | that dropped down a tree ... would be there a person it would take him down as well |
| 02:13.05 | cheasee | you were talking? dont lie, i saw what you did: http://www.detailverliebt.de/bilder/geniale-bildbearbeitung-von-erik-johansson.jpg |
| 02:13.33 | cheasee | ;) |
| 02:14.22 | gusto | btw. when that robot puts that cables inside the shitter tube, where do the ppl shit during that time? |
| 02:14.55 | gusto | i would not want A1 to tell me when to and when not to use my loo |
| 02:14.56 | ChannelZ | in plants |
| 02:15.04 | cheasee | it gets routed over other connections :P |
| 02:15.44 | cheasee | dont you know dynamic routing? |
| 02:16.00 | gusto | no |
| 02:16.30 | WIMPy | It's a misery. |
| 02:16.32 | gusto | i have only one route to my loo |
| 02:16.54 | WIMPy | That's what you think. |
| 02:17.00 | cheasee | sure, you also got only one route to your modem |
| 02:17.11 | gusto | yes |
| 02:17.32 | cheasee | WIMPy: lol :D |
| 02:18.17 | gusto | so |
| 02:21.51 | gusto | someday i ll have an SDSL ;-) |
| 02:22.22 | gusto | i am sick and tired of this asymetric connections |
| 02:29.34 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 02:42.07 | WIMPy | FTTH |
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| 03:03.55 | gusto | N/A |
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| 12:19.53 | hugo | [Jan 6 12:19:14] WARNING[-1]: chan_sip.c:4374 __sip_autodestruct: Autodestruct on dialog '7b6d6dc63519577827020b3a6e486300@asterisk_ip:5061' with owner SIP/myuser-0000000d in place (Method: INVITE). Rescheduling destruction for 10000 ms |
| 12:20.20 | hugo | has been going on all night, no clients connected to it, any ideas what could be causing it? |
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| 12:41.28 | hugo | ok guys I think I may be on to something here regarding the issues with nat clients.. I turned SIP debug on and now I see Transmitting (no NAT) to (...) in one of the clients |
| 12:41.33 | hugo | which is also behind a NAT |
| 12:41.51 | hugo | shouldn't it always say Transmitting (NAT) ? |
| 12:42.16 | hugo | we're running diff uas and the other party is on tcp/tls while I'm connected via udp but that shouldn'tmake a difference here? |
| 12:48.09 | hugo | and what's this stuff with "reliably transmitting (NAT)", how does it differ from "transmitting (nat)" ? |
| 12:51.55 | ghost75 | different nat modes |
| 12:53.24 | ghost75 | Asterisk 1.8: |
| 12:53.24 | ghost75 | The 'nat' option has now been been changed to have yes, no, force_rport, and comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the remote side requests it and disables symmetric RTP support. Setting it to force_rport forces RFC 3581 behavior and disables symmetric RTP support. Setting it to comedia enables RFC 3581 behavi |
| 12:54.37 | hugo | hm |
| 12:55.19 | hugo | in sip.conf general I have nat=force_rport,comedia now |
| 12:55.33 | hugo | and in the others configuration just nat=yes |
| 12:55.38 | hugo | others=users |
| 12:57.14 | hugo | maybe I got that part wrong too.. |
| 13:01.12 | ghost75 | i think nat should be only under general or sip peer |
| 13:02.04 | hugo | I'll read some more about it |
| 13:02.16 | hugo | thought I read somewhere that the directmedia etc settings only apply if client has nat=yes too |
| 13:02.59 | hugo | at any rate is it correct that for clients behind nat one would always need comedia,force_rport (if client didn't set rport) and directmedia=no? |
| 13:04.24 | ghost75 | in directmedia the endpoints talk directly to each other |
| 13:05.15 | hugo | directmedia=no is supposed to prevent that |
| 13:05.59 | hugo | which I think would always be required with NAT involved because the clients won't be able to reach their local NAT IPs and the port that asterisk is using to go back to them will appear closed to any other IP |
| 13:07.18 | ghost75 | yes i think directmedia is when every phone has its own public ip address |
| 13:08.24 | ghost75 | i have nat=no and no port forwardings on router |
| 13:09.06 | hugo | yeah, I reckon my problems are coming from the fact that I have NAT on asterisk and everyone connecting to it is behind NAT as well |
| 13:09.49 | ghost75 | do you have all phones in same lan as asterisk? |
| 13:10.43 | hugo | nope |
| 13:10.55 | hugo | asterisk in the US laptops connecting from all over |
| 13:11.00 | hugo | all of them under nat as well |
| 13:17.05 | ghost75 | can you portscan me? |
| 13:17.15 | ghost75 | i am curious if 5060 is open or not |
| 13:22.31 | hugo | sure |
| 13:22.33 | hugo | what's ur ip? |
| 13:24.33 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 13:24.54 | hugo | ghost75 |
| 13:26.53 | ghost75 | IPv4: 88.66.185.223 |
| 13:27.06 | ghost75 | udp not tcp |
| 13:28.28 | hugo | it's there |
| 13:28.39 | ghost75 | open? |
| 13:28.41 | hugo | I nc'd and wrote something |
| 13:28.43 | hugo | it closes the connection |
| 13:29.45 | ghost75 | i think asterisk uses some tricks like skype |
| 13:30.11 | hugo | it actually looks closed, nevermind |
| 13:30.21 | hugo | pointed baresip to it, I can only see outbound packets |
| 13:30.22 | hugo | 0 reply |
| 13:30.35 | hugo | ua: dude@88.66.185.223: Register: Operation timed out |
| 13:30.50 | ghost75 | good |
| 13:31.23 | hugo | hey I gotta go now |
| 13:31.25 | hugo | be back soon |
| 13:46.32 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 14:11.34 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:12.15 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.82.52) |
| 14:26.36 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 14:51.15 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 14:51.26 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:04.25 | hugo | after killing asterisk not so nicely and starting it up again I see ERROR[-1] tcptls.c: Unable to connect SIP socket to 62.195.xxx.xxx:4491: Connection refused repeatedly |
| 16:04.42 | hugo | like it still remembers the client's address/port from a previous session |
| 16:04.54 | hugo | how do I convince it to stop trying ? |
| 16:15.27 | *** join/#asterisk shadar (~eugene@37.113.202.81) |
| 16:16.37 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 16:17.55 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 16:19.16 | danfromuk | Hi, i'm struggling with DTMF and asterisk. Usually I do things over sip and know how to deal with that. I've got a client using ISDN lines for calls and asterisk doesnt want to respond to DTMF signals when doing things like READ. Any ideas what could be the cause? |
| 16:20.09 | WIMPy | What channeltype? |
| 16:21.20 | danfromuk | mIsdn |
| 16:21.22 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 16:23.28 | WIMPy | That direction should work right out of the box. |
| 16:24.55 | danfromuk | hmm. thats what i thought |
| 16:25.20 | danfromuk | exit |
| 16:25.23 | danfromuk | oops |
| 16:26.09 | danfromuk | Just to check, this is the correct syntax for asterisk 1.4? Read(result|divert-menu-1|1) |
| 16:26.20 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 16:27.49 | [TK]D-Fender | danfromuk: Show us the full call, and you still shouldn't be using "|" as a delimiter |
| 16:28.12 | danfromuk | Its an old asterisk 1.4 installation. The client doesnt want to upgrade. |
| 16:28.48 | WIMPy | | was for 1.2 and earlier. |
| 16:32.03 | danfromuk | Ok, sorry, i thought it was changed in 1.6 |
| 16:34.33 | danfromuk | However, changing it to comma didnt help. Call log here http://pastebin.com/1KiGMx17 |
| 16:34.38 | danfromuk | Dial plan following in a moment. |
| 16:36.39 | danfromuk | Dialplan http://pastebin.com/VB7bSnMp |
| 16:36.51 | danfromuk | Caller is dialling 7204408 |
| 16:37.07 | danfromuk | External caller |
| 16:37.21 | danfromuk | Or 204408 |
| 16:37.29 | danfromuk | I didnt make most of this plan. I've just taken it over. |
| 16:46.05 | danfromuk | I dont think its a dialplan issue though. |
| 16:47.56 | danfromuk | Ah, it seems it was an issue with my home phone. Some reason it didnt like my dtmf tones. |
| 16:48.00 | danfromuk | Mobile works fine. |
| 16:48.28 | danfromuk | Thanks for your help. |
| 16:48.59 | davlefou | hi, what is use the regexten? Is for register line? |
| 16:51.04 | [TK]D-Fender | davlefou: It's pretty much worthless. |
| 16:51.43 | davlefou | Thanks |
| 16:58.41 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 17:01.49 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 17:05.34 | WIMPy | danfromuk: You can tune the DSP module if you feel like it. |
| 17:10.26 | *** join/#asterisk Tim_Toady (~fuzzy@194.50.55.245) |
| 17:11.46 | *** join/#asterisk vlad_starkov (~vlad_star@wn1nat15.beelinegprs.ru) |
| 17:13.11 | *** join/#asterisk vlad_sta_ (~vlad_star@wn1nat15.beelinegprs.ru) |
| 17:16.45 | *** join/#asterisk vlad_st__ (~vlad_star@wn2nat46.beelinegprs.ru) |
| 17:26.31 | *** join/#asterisk vlad_sta_ (~vlad_star@83.149.8.185) |
| 17:29.36 | hugo | if in sip show channelstats one of the peers shows as the local (NATted) IP, what could be wrong? |
| 17:29.43 | hugo | in sip show peers I see the external IP on both |
| 17:35.57 | ghost75 | wtf, when i park a call then the parking number is announced to the other channel |
| 17:40.57 | hugo | Peer audio RTP is at port: local_nat_ip:some_port |
| 17:41.10 | hugo | I suppose that's wrong .. |
| 17:41.23 | hugo | if it means what I think it means, that it's actually using the internal IP to address the rtp stream |
| 17:42.45 | ghost75 | once i had problem with my router, it had old nat table and asterisk was using local ip instead |
| 17:44.57 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:45.10 | ghost75 | why asterisk begins already with transfer when i press only # |
| 17:45.31 | ghost75 | in features.conf there is not just only # |
| 17:56.43 | [TK]D-Fender | ghost75: Show us. |
| 18:06.07 | ghost75 | dtmf tx method: auto |
| 18:06.11 | ghost75 | dtmf tx mode: strict |
| 18:06.17 | ghost75 | thats on cisco ata |
| 18:10.06 | [TK]D-Fender | the * side... |
| 18:10.18 | [TK]D-Fender | Your configs. You call. |
| 18:10.43 | ghost75 | do see any setting here worth changing: https://www.sugarsync.com/piv/D234635_65941903_648551 |
| 18:13.06 | [TK]D-Fender | ghost75: * is clearly getting your DTMF. It is starting a transfer. Stop staring at your ATA and start looking at * |
| 18:13.33 | ghost75 | features.conf? |
| 18:13.38 | [TK]D-Fender | That. |
| 18:13.40 | [TK]D-Fender | And your dialplan. |
| 18:13.43 | [TK]D-Fender | And the call debug |
| 18:13.48 | [TK]D-Fender | PASTEBIN. |
| 18:16.26 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.185) |
| 18:16.36 | ghost75 | http://pastebin.com/UghCNcdu |
| 18:16.46 | ghost75 | that says also default would be # |
| 18:18.13 | ghost75 | do i need to enable features.conf file somewhere? |
| 18:19.33 | [TK]D-Fender | Where is the rest? |
| 18:19.51 | ghost75 | i think features.conf is just not used |
| 18:24.21 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-195-147.red.bezeqint.net) |
| 18:24.28 | ghost75 | http://pastebin.com/iEv8mhCw |
| 18:25.44 | ghost75 | dial cmd is: |
| 18:25.45 | ghost75 | exten => _1X,1,Dial(SIP/${EXTEN},20,tTkK) |
| 18:27.31 | ghost75 | when i press # then i see in cli: Started music on hold, class 'default', on SIP/12-0000009a |
| 18:28.39 | ghost75 | http://pastebin.com/m1GB2med |
| 18:33.58 | [TK]D-Fender | ghost75: Did you restart * since having made that change? "ls -la /et/asterisk" > PB |
| 18:35.33 | ghost75 | i changed only dialplan and did reload dialplan but can restart all |
| 18:36.31 | [TK]D-Fender | So... you didn't even apply your changes from features.conf defaults... |
| 18:37.34 | ghost75 | fuuuuuuuuuuuuuuuuuuu |
| 18:37.53 | WIMPy | ck? |
| 18:37.57 | ghost75 | features.conf i changed yesterday, looks like i forgot to restart yes |
| 18:38.31 | WIMPy | Didn't I ask if it was activated? |
| 18:39.04 | ghost75 | u did and i thought i did |
| 18:39.57 | [TK]D-Fender | "Think" .... http://i1.kym-cdn.com/photos/images/newsfeed/000/340/205/4d5.png |
| 18:43.31 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
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| 19:35.57 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
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| 20:02.33 | hugo | my friend tries to place a call and soon after asterisk boots him off.. seems to happen pretty reliable |
| 20:02.33 | verywiseman | if i want to send a call through ATA , which one syntax is true : Dial(sip/ATA/Extension) , or Dial(sip/Extension@ATA) , where ATA is defined in sip.conf? |
| 20:02.34 | hugo | bly |
| 20:02.42 | hugo | says peer becomes unreachable |
| 20:03.23 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 20:03.30 | hugo | ideas? |
| 20:03.52 | WIMPy | verywiseman: Preferrably the first. |
| 20:05.16 | [TK]D-Fender | verywiseman: NEITHER |
| 20:05.26 | [TK]D-Fender | verywiseman: Dial(SIP/peer) |
| 20:07.19 | WIMPy | If it has only one port. |
| 20:07.50 | verywiseman | [TK]D-Fender, would you please explain more about SIP/peer |
| 20:07.56 | [TK]D-Fender | most multiport ATA's us separate SIP ports on the same IP. |
| 20:08.03 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:08.09 | [TK]D-Fender | You don't dial a target extension at them |
| 20:08.17 | [TK]D-Fender | I have never seen any that you do this for |
| 20:08.47 | WIMPy | Many contain their own little PBX. |
| 20:09.10 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 20:09.22 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 20:09.24 | [TK]D-Fender | WIMPy: Which? |
| 20:09.59 | WIMPy | All those low cost things that also contain a router and modem and whatnot. |
| 20:11.06 | hugo | I set directmedia back to yes, me and my friend are on the same LAN (asterisk is behind NAT remote) |
| 20:11.17 | hugo | sip show channelstats shows our internal-nat IPs |
| 20:11.24 | hugo | when I call him I can see him but he can't see me (we're in the same room) |
| 20:11.27 | hugo | no firewalls in our laptops |
| 20:11.32 | hugo | what on earth can possibly be wrong? :( |
| 20:12.42 | [TK]D-Fender | WIMPy: None of the linksys/ciso, Grandstream ones... |
| 20:12.55 | [TK]D-Fender | WIMPy: Not Audiocode from what I've seen.... |
| 20:13.05 | [TK]D-Fender | WIMPy: Got a sample that people in here have really used? |
| 20:13.32 | verywiseman | [TK]D-Fender, ok , how can i send extension ? |
| 20:13.41 | [TK]D-Fender | verywiseman: You don't |
| 20:15.43 | verywiseman | [TK]D-Fender, sorry , i didn't understand , what is Dial() syntax to generate a call through ATA to PSTN? |
| 20:15.58 | [TK]D-Fender | verywiseman: You are using the wrong term.... |
| 20:16.01 | [TK]D-Fender | ~ata |
| 20:16.02 | infobot | methinks ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 20:16.05 | [TK]D-Fender | TERMINAL. |
| 20:16.08 | [TK]D-Fender | A PHONE is a terminal. |
| 20:16.10 | [TK]D-Fender | not a LINE |
| 20:16.34 | [TK]D-Fender | You are using "SIP gateway", and as usual your description was vague |
| 20:16.57 | [TK]D-Fender | verywiseman: Dial(SIP/peer/numbertodial) is the normal way for that |
| 20:21.17 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 20:22.43 | hugo | has anyone managed to get 2 way audio+video at all with all clients behind nat and asterisk on another nat ? |
| 20:24.03 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 20:24.33 | volga629 | Hello Everyone, trying troubleshoot http://fpaste.org/Ydxw/ Got SIP INFO response 415 "Unsupported Media Type" back from host '10.140.254.75:5060' |
| 20:24.40 | volga629 | any help thank you |
| 20:45.45 | ghost75 | unsupported codec? |
| 20:50.25 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 21:46.38 | ChannelZ | volga629: do you have g729 licenses? |
| 21:47.23 | ChannelZ | you also seem to be having some other network issues with these retransmits |
| 21:50.23 | *** join/#asterisk ruied (~ruied@po-217-129-254-134.netvisao.pt) |
| 22:08.08 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
| 22:16.01 | *** join/#asterisk gg608f (~Adium@187.207.6.250) |
| 22:16.27 | *** join/#asterisk mjordan (~mjordan@216.186.152.216) |
| 22:16.27 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:43.47 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 23:00.33 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:01.18 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:03.32 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 23:06.11 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:09.38 | volga629 | yes I have lince |
| 23:09.47 | volga629 | yes I have license |
| 23:11.26 | volga629 | hmmmmmmmmm -- Channels: 32541 (incorrect host-id) |
| 23:12.01 | volga629 | why is that over the sudden, I din't touch server at all |
| 23:12.19 | volga629 | might be re transmit from this too |
| 23:13.15 | volga629 | this l2tp ipsec client might be mtu issue |
| 23:16.05 | volga629 | license can screw up only if nic mac was change right ? |
| 23:18.12 | volga629 | uffffffff Server response: 469 - Maximum Activations Reached. Please contact Digium Support |
| 23:21.52 | volga629 | transmitting might firewall issue |
| 23:24.33 | volga629 | ChannelZ, ghost75. thank you for help. yes always need look on simple thinks first :-) |
| 23:30.04 | *** join/#asterisk mbrit (~mbrit@186.120.91.196) |
