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| 01:59.17 | signpost | would anyone know a good voip provider with incoming sms? |
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| 02:35.42 | atan | Okay maybe I am totally nuts but, err, No such command 'sip reload' (type 'core show help sip reload' for other possible commands) |
| 02:36.23 | atan | There must be another config file I am missing |
| 02:36.40 | atan | I'm trying to start off plain without the 100 or so config samples |
| 02:41.12 | psykon | atan: why not just use 'reload' |
| 02:41.32 | WiretapMac | psykon: because 'reload' is a 1.6 command |
| 02:42.06 | atan | Hmm, must require modules.conf |
| 02:42.55 | atan | Okay I'm almost stumped. Which config files are required by asterisk? |
| 02:44.56 | atan | See on one server I have these: http://pastebin.com/KpGCKY48 |
| 02:45.06 | atan | I could be wrong, but there's clearly more than needed in that bloody thing |
| 02:45.54 | atan | My other box has just sip.conf modules.conf and asterisk.conf |
| 02:54.54 | atan | Hmm okay seem to be good to go. Had to tell it to load chan_sip.so but I don't believe I did previously, it was commented out in the old config... but autoload = on was in there. Oh well. Working now. |
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| 03:37.37 | atan | Which modules must be loaded for your extensions.conf dialplan to load? |
| 03:40.44 | atan | Meh, pbx_config.so I suppose. |
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| 04:11.35 | atan | Okay so if autoload=yes phone connects fine, but using http://pastebin.com/7Su6CtXD does me nothing and shows 'number unavailable' on the phone trying to call. In the console I don't even see the attempt being made. |
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| 04:33.02 | atan | Okay my bad. Requires bridge_builtin_features.so but I didn't find any reference to that on the voip-info site. Darn :D anyway, getting somewhere here now! |
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| 07:58.02 | irroot | goodmorning all |
| 07:59.15 | irroot | finishing off a script to create a CA for VOIP with a CA cert signed with the system CA and create a LDAP entry for each phone for TLS |
| 07:59.38 | irroot | have not used TLS + * before anyone done this |
| 08:00.17 | irroot | way too much fraud out in the wild want to verify identity with CRL |
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| 09:23.11 | bintut | waves |
| 09:23.21 | bintut | anyone familiar on this? => WARNING[20619]: res_srtp.c:338 ast_srtp_unprotect: SRTP unprotect: authentication failure (and) WARNING[20618]: app_dial.c:2041 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
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| 09:23.37 | bintut | on the same private LAN where the Bria for iPhone4 (wireless) tries to call the SNOM 300 (wired), i am getting this error message which resulted for me to hear a busy tone => app_dial.c:2041 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 09:23.43 | bintut | but calling the other way around, meaning, calling from SNOM 300 to Bria for iPhone4 works |
| 09:25.57 | irroot | bintut yeah you need SRTP setup proprly |
| 09:26.13 | irroot | the "encryption = yes" in sip .conf |
| 09:27.01 | irroot | and be aware snom300 needs its auth taglen set to 80 |
| 09:27.30 | irroot | i have a patch that allows use of 32 bit authtag's and optional SRTP |
| 09:27.41 | bintut | irroot: yup, i have it on my sip.conf but it only happens with the bria for iphone. yesterday, i was able to change the aes_32 to aes_80 and that's why my snom 300 works.. now with bria for iphone that it doesn't have a way to configure it, i'm not sure how to make it work |
| 09:28.15 | irroot | https://reviewboard.asterisk.org/r/1173/ |
| 09:28.37 | irroot | set encryption = try |
| 09:28.59 | irroot | with the patch and make snom optional in the phone |
| 09:29.36 | irroot | or turn srtp off |
| 09:31.18 | irroot | you can sip debug the bria and see what SDP is offered |
| 09:31.36 | irroot | is there a AVP or SAVP ?? |
| 09:32.32 | bintut | irroot: i assume that your patch has not yet applied in asterisk-1.8.4.1, right? |
| 09:32.52 | irroot | no it works in 1.8.4 but its for trunk |
| 09:32.58 | bintut | what is avp or savp? sorry.. |
| 09:32.58 | irroot | i have it in SVN |
| 09:33.05 | irroot | in the sip debug |
| 09:33.34 | irroot | when a device uses SRTP it uses SAVP to indicate this AVP is with no encryption |
| 09:34.00 | bintut | wait.. let me paste the sip messages.. |
| 09:36.20 | irroot | unless you have to use SRTP turn it off |
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| 09:39.28 | bintut | irroot: kindly check this => http://www.pastie.org/1988150 |
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| 09:39.39 | bintut | irroot: i'm trying to make srtp work |
| 09:40.22 | cHarNe2 | what codec should i use for sip-clients on mobilephones? |
| 09:41.09 | ectospasm | cHarNe2: is bandwidth a concern? |
| 09:41.17 | bintut | irroot: and also, since i upgraded to asterisk-1.8.4.1, i had weird problems in using analog phones with dahdi |
| 09:41.21 | irroot | charne2 as narrow as possible gsm/g72[39]/ilbc |
| 09:41.34 | cHarNe2 | ectospasm: im just using standard now and i have a lot of delay |
| 09:42.29 | ectospasm | standard... meaning what, exactly? |
| 09:42.40 | cHarNe2 | ectospasm: i havent changed from the default install |
| 09:42.48 | ectospasm | cHarNe2: default doesn't compute |
| 09:42.54 | ectospasm | afaik there is no default |
| 09:43.10 | ectospasm | unless you mean G.711(u|a) |
| 09:43.31 | cHarNe2 | okay |
| 09:43.55 | ectospasm | cHarNe2: what have you explicitly set in disallow/allow lines in sip.conf? |
| 09:45.11 | cHarNe2 | disalow all; allow ulaw |
| 09:45.29 | cHarNe2 | so im using ulaw? |
| 09:46.08 | cHarNe2 | it will all be sipclients in cellphones, shold i use the 'gsm'? |
| 09:46.51 | phpboy | cHarNe2: you connecting over 3g? |
| 09:47.20 | cHarNe2 | phpboy: yes for now |
| 09:47.32 | irroot | bintut cant be 100% sure |
| 09:47.38 | cHarNe2 | phpboy: should i use other if on wifi? |
| 09:48.08 | irroot | why you getting auth denied should not |
| 09:48.19 | ectospasm | cHarNe2: I'd recommend a compressed codec, like gsm, G.722, or G.729 (listed in order of increasing performance) |
| 09:48.38 | ectospasm | actually, gsm may be better than G.722, dunno |
| 09:48.45 | phpboy | I have something similar going in my home country... on the gsm network use gsm or if it makes financial sense G729 |
| 09:48.46 | ectospasm | I never come across G.722... |
| 09:48.53 | bintut | irroot: what do you mean? sorry.. |
| 09:49.11 | irroot | in the trace there is a auth problem get error 40 |
| 09:49.11 | phpboy | on wifi ulaw is fine |
| 09:49.13 | irroot | 401 |
| 09:49.14 | cHarNe2 | kk, ill use gsm then |
| 09:49.28 | phpboy | cHarNe2: gsm will be the most straight forward |
| 09:49.30 | *** part/#asterisk jqb (~jqb@85.26.184.164) |
| 09:49.44 | phpboy | and it's tried and tested on gsm networks |
| 09:49.54 | irroot | line 48 |
| 09:49.56 | phpboy | cHarNe2: what type of phone do you have you sip client installed on? |
| 09:50.10 | cHarNe2 | android's and ihpne's |
| 09:50.19 | phpboy | I've got mine on iphone |
| 09:50.22 | phpboy | works like a charm |
| 09:50.28 | irroot | the 3G networks arround here shape and drop voip so as low as possible is all that works |
| 09:50.35 | cHarNe2 | ok, what app u using? |
| 09:51.03 | phpboy | good question, I'll have to check... |
| 09:51.18 | cHarNe2 | :) |
| 09:52.04 | phpboy | I've played around with two |
| 09:52.36 | phpboy | the reason I didn't stick with the first is because if a GSM call came through will you were on a SIP call it would automatically put the sip call on hold |
| 09:52.42 | phpboy | which is extremely annoying |
| 09:53.38 | cHarNe2 | ok |
| 09:56.11 | ectospasm | I'd imagine that'd depend on the carrier. |
| 09:56.22 | cHarNe2 | phpboy: what was the name of the app? |
| 09:56.25 | ectospasm | as the SIP connection is simply a data connection |
| 09:57.05 | *** join/#asterisk rethus (~suther@p549A7BCF.dip.t-dialin.net) |
| 09:57.35 | rethus | on asterisk-cli i got CDR Variables if i do "core show channel SIP/000000bc |
| 09:57.45 | rethus | all those have prefix level1: |
| 09:57.57 | rethus | what sense does this level1 have? |
| 09:58.04 | rethus | which other states are available |
| 09:58.14 | ectospasm | rethus: your question lacks context |
| 09:58.42 | rethus | i try to unserstand what for this part of show channel is |
| 09:58.42 | ectospasm | "core show channel <channel>" won't show CDR variables IIRC |
| 09:58.51 | ectospasm | rethus: pastebin |
| 10:00.00 | bintut | irroot: isn't it because of line 152? |
| 10:00.14 | rethus | http://pastebin.com/3qN12s1C, |
| 10:01.27 | ectospasm | rethus: level1 is the main CDR level, all of those fields look like what populate one CDR entry |
| 10:01.40 | bintut | irroot: the brian for iphone was able to register and able to perform the first few steps in the sequence but then when it reached the step 4 which is the actual dial, it got that line 152 message |
| 10:01.43 | ectospasm | ...I don't know if I've ever seen a level2 entry |
| 10:02.26 | rethus | can you explain what exactly CDR mean? |
| 10:02.42 | bintut | irroot: maybe the reason why i am hearing a busy tone right away |
| 10:02.55 | ectospasm | rethus: CDR == Call Detail Record |
| 10:02.59 | bintut | without even the snom 300 to ring |
| 10:03.01 | irroot | binut yip its been rejected almost right away error 401 |
| 10:03.12 | rethus | ectospasm: k. thanks |
| 10:03.41 | irroot | as if it does not exist and with the authreject that 1.8 defaults to you will get 401 |
| 10:05.03 | bintut | irroot: well, this only happens on my bria for iphone.. |
| 10:05.08 | rethus | ectospasm: can i set CDR-Variable ? |
| 10:05.23 | ectospasm | rethus: you can, gimme a sec and I'll look it up. |
| 10:05.34 | rethus | ectospasm: k, thanks |
| 10:05.39 | phpboy | ectospasm: what I mean is if you were on a sip call and you got a call from the gsm network the client would put the sip call on hold in favour of the gsm call :\ |
| 10:05.47 | bintut | irroot: the only thing that i can't confirm from the bria for iphone is the cipher.. for snom 300, i set it to aes_80.. |
| 10:06.08 | cHarNe2 | my sipclient has a function to send mesages, is that possimle over an asterisk? |
| 10:06.34 | phpboy | what type of messages? |
| 10:06.39 | irroot | it sets both cant tell you off hand what happens in that case |
| 10:06.43 | rethus | in my pastebin u see above CDR that i set own-one... like PINENTRY, but wouldbe nice, if i could push it to the CDR-section |
| 10:06.59 | cHarNe2 | phpboy: i have no idéa :P |
| 10:07.01 | irroot | have to check source |
| 10:07.09 | irroot | when i work on it ill check |
| 10:07.11 | cHarNe2 | :S well ill guess ill skip that for now |
| 10:07.52 | phpboy | :P |
| 10:08.41 | rethus | found this: https://wiki.asterisk.org/wiki/display/AST/CDR+Variables |
| 10:08.59 | bintut | irroot: ok. thanks again. ;) |
| 10:09.05 | ectospasm | rethus: you can use the CDR() function to set a specific CDR variable in dialplan. See "core show function CDR" |
| 10:09.20 | rethus | k, i'll try it |
| 10:12.46 | ectospasm | rethus: you may want to consider CEL rather than CDR... |
| 10:12.56 | ectospasm | (CEL==Channel Event Logging) |
| 10:14.06 | WiretapSeven | cHarNe2, no, it is not possible outside a call |
| 10:14.13 | rethus | i use phpagi - but i don't see, how to set(CDR()) |
| 10:14.31 | ectospasm | rethus: I dunno about AGI, sorry |
| 10:16.49 | cHarNe2 | WiretapSeven: ok, ty |
| 10:16.59 | *** part/#asterisk cHarNe2 (thorn@gateway/shell/bshellz.net/x-rvqclxbfvnkovbxm) |
| 10:17.34 | rethus | whats the benefit of CEL? |
| 10:23.06 | rethus | Anyone know, how i can use Set(CDR(name)=value) via AGI ? |
| 10:26.11 | tzafrir_laptop | exec? |
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| 10:42.44 | rethus | tzafrir_laptop: with which command? i didn't get it to work |
| 10:43.28 | phpboy | rethus: give an example of exactly what you're trying to do so that we can help you find the best solution |
| 10:43.42 | tzafrir_laptop | exec Set(CDR(name)=value) |
| 10:43.45 | tzafrir_laptop | Or something similar |
| 10:48.03 | rethus | tzafrir_laptop: i try this, tdidn't work |
| 10:48.38 | rethus | phpboy: in my pastebin http://pastebin.com/3qN12s1C, u see above CDR that i set own-one... like PINENTRY, but wouldbe nice, if i could push it to the CDR-section |
| 10:48.50 | rethus | itry to set own CDR-Vars via agi |
| 10:48.53 | rethus | phpagi |
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| 11:03.17 | irroot | ok my CA for snom phones auto provisioning working |
| 11:03.36 | irroot | any idea how to set TLS cert on polycoms ?? |
| 11:05.59 | irroot | trick is to make sure the private keys are only exposed to the phone and only from the phones registed ip the first time round |
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| 11:27.53 | Roy123423 | I have got 2 trunks (sip1 and sip2), I would like to forward the incoming call to user1 if from sip 1 and so on. How would I go about this? I could not see any option on my inbound route to allow this. Thanks for all help in advance |
| 11:28.29 | ectospasm | Roy123423: there is the Transfer() dialplan application. |
| 11:32.01 | *** join/#asterisk denysonique (~quassel@unaffiliated/dennisonicc) |
| 11:32.12 | Roy123423 | ectospasm: Looking at this now, will reply soon. |
| 11:32.22 | kaldemar | Roy123423: #freepbx |
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| 11:32.47 | denysonique | Hello phreakers |
| 11:33.09 | kaldemar | it is done by setting contexts in sip.conf |
| 11:33.38 | otwieracz | Hello. |
| 11:34.06 | otwieracz | I'm trying to setup Aterisk behind NAT with SipGate(.co.uk) |
| 11:34.11 | otwieracz | I forwarded: |
| 11:34.17 | otwieracz | 5060 UDP |
| 11:34.24 | otwieracz | udp dpts:10000:10050 to:10.100.0.5 |
| 11:34.29 | otwieracz | In my rtp.conf: |
| 11:34.33 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 11:34.39 | otwieracz | rtpstart=10000 |
| 11:34.41 | otwieracz | rtpend=10050 |
| 11:35.09 | otwieracz | I can do outgoing calls (I succesfully conected to BT information service) |
| 11:35.25 | otwieracz | But when someone is calling me he's getting âwrong numberâ. |
| 11:35.31 | otwieracz | What should I do? |
| 11:36.04 | *** join/#asterisk jhmichelle352 (~jhmichell@41.69.251.153) |
| 11:36.07 | otwieracz | I don't see in Asterisk log anything about that incoming call (asterisk -vvvf). |
| 11:36.17 | *** join/#asterisk vfabi (~fabi@host-static-188-237-240-194.moldtelecom.md) |
| 11:38.21 | angryuser_laptop | otwieracz, asterisk -rvvvvvv |
| 11:38.51 | angryuser_laptop | otwieracz and o a sip debug on oyur provider when calling |
| 11:39.30 | atan | Hmm, once I connect a call to a SIP extension is there a way to listen for "##" being pressed by the called party? |
| 11:39.43 | atan | If they press ## I want to run an AGI |
| 11:39.45 | *** join/#asterisk rethus (~suther@p549A7BCF.dip.t-dialin.net) |
| 11:41.46 | otwieracz | I'm watching iptables rules at router, but nothing is coming. |
| 11:42.08 | otwieracz | Those rules counters doesn't change when somebody is tryging to call to my SIP number. |
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| 12:01.21 | atan | Well I'm stumped. SIP phone is saying 'number unreachable' but asterisk isn't even showing the dial attempt |
| 12:01.32 | atan | Phone appears registered, shows in sip show peers |
| 12:02.04 | otwieracz | I have the same. |
| 12:02.25 | atan | otwieracz, are you running the latest? |
| 12:02.41 | otwieracz | Asterisk 1.8.4 |
| 12:02.44 | atan | .1? |
| 12:02.54 | otwieracz | [root@linvoip asterisk]# asterisk -V |
| 12:02.55 | otwieracz | Asterisk 1.8.4 |
| 12:03.15 | atan | Wait a sec. |
| 12:03.18 | atan | Are you at linode? |
| 12:03.29 | atan | Just, err, 'linvoip' makes me wonder. |
| 12:03.42 | otwieracz | linvoip, my hostname. |
| 12:04.46 | atan | Well if we have the same issue stick around O_O |
| 12:04.56 | otwieracz | sipgate? |
| 12:05.24 | atan | Nah, voipms but the issue is between device and server. Not trying to call outbound |
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| 12:23.50 | atan | make menuselect seems to pull what I've previously used in the last install. Is there a way to reset that? I clearly goofed something up when I unselected everything. |
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| 12:27.38 | atan | oh I'm full of it. Works now. |
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| 13:25.34 | Dovid | anyone know of a phone that will re-invite after X amount of time ? |
| 13:26.57 | bintut | waves.. gtg now.. thanks.. ;) |
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| 13:36.05 | Dovid | anyone know of a phone that will re-invite after X amount of time ? |
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| 15:28.48 | *** join/#asterisk phigan (446da341@gateway/web/freenode/ip.68.109.163.65) |
| 15:28.52 | phigan | hi |
| 15:29.55 | phigan | if i have a line like this: exten => _8NXXNXXXXXX,1,AGI(script.agi) .. how can I remove the 8 before passing the number to the script? |
| 15:30.34 | WIMPy | ${EXTEN:1} |
| 15:31.39 | phigan | and where's the ${EXTEN:1} go? |
| 15:32.08 | phigan | if I had an ${EXTEN} in the line, I would've added :1 :) |
| 15:32.19 | WIMPy | Where you want to pass it to your scrpit. |
| 15:32.34 | WIMPy | So how do you pass the EXTEN then? |
| 15:32.48 | phigan | i just have this line: exten => _8NXXNXXXXXX,1,AGI(script.agi) |
| 15:32.55 | phigan | the number gets passed to script.agi |
| 15:33.22 | WIMPy | Then you have to do it in the script. |
| 15:34.16 | phigan | would exten => _8NXXNXXXXXX,1,AGI(script.agi, ${EXTEN:1}) work, maybe? |
| 15:35.26 | silkcut | well i guess (you can get the result as ${arg1} |
| 15:35.35 | silkcut | http://www.voip-info.org/wiki/view/Asterisk+AGI |
| 15:35.41 | phigan | that's what I'm reading |
| 15:35.52 | phigan | but I can't tell if having nothing at all means the # is arg1 |
| 15:37.35 | WIMPy | Nothing is nothin. |
| 15:38.32 | phigan | it just knows? :) |
| 15:39.07 | WIMPy | No magic |
| 15:39.36 | WIMPy | It probably reads EXTEN itself. |
| 15:44.15 | irroot | the agi probably looks at the exten |
| 15:44.21 | irroot | it does not need it passed |
| 15:44.58 | irroot | you could do a _8...,1,Goto(${EXTEN:1}) |
| 15:45.11 | irroot | then have _XXXX....,n,AGI |
| 15:45.34 | irroot | editing the agi script is a alternative |
| 15:45.56 | irroot | or passing it in script.agi?exten= |
| 15:46.01 | irroot | similar to http |
| 15:46.26 | irroot | but the script needs to parse and process |
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| 15:55.48 | phigan | hm |
| 15:56.15 | phigan | i'd like to not have to modify the script. are you saying that script.agi?exten= would require that? |
| 15:57.13 | WIMPy | Yes |
| 15:57.36 | WIMPy | You have to go the goto way then. |
| 15:58.17 | phigan | bummr. also, if I already have a exten => _NXXNXXXXXX,etc .. the exten => _8NXXNXXXXXX,1,Goto(${EXTEN:1}) would go to that.. |
| 15:58.29 | phigan | wouldn't it? |
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| 16:15.22 | phigan | eh, i changed it in the agi.. i just have to remember to change it there if the agi ever gets updated |
| 16:15.49 | phigan | thanks for the help, gents. |
| 16:21.58 | *** join/#asterisk vfabi (~fabi@host-static-188-237-241-39.moldtelecom.md) |
| 16:23.01 | phigan | here's something i can't figure out.. |
| 16:23.26 | phigan | if i'm on the phone with someone, and the other end hangs up first, all my phones suddenly ring with the callerid of the extension i'm on |
| 16:23.42 | phigan | is it because i'm missing a hangup in the dialplan or something? |
| 16:49.22 | *** join/#asterisk irroot (~irroot@41.49.85.192) |
| 16:53.17 | phigan | irroot: happen to see my question about extensions ringing when the other end hangs up first? |
| 16:53.29 | irroot | sorry got bounced |
| 16:53.33 | irroot | what up |
| 16:53.36 | phigan | if i'm on the phone with someone, and the other end hangs up first, all my phones suddenly ring with the callerid of the extension i'm on |
| 16:53.47 | phigan | i'm guessing i'm missing something in my DP |
| 16:53.54 | irroot | yeah 4 sho |
| 16:54.05 | phigan | i have a HangUp() at the end.. |
| 16:54.07 | irroot | set verb 3 |
| 16:54.20 | phigan | err HangUp |
| 16:54.26 | irroot | and look at the dial command does it have a g option ?? |
| 16:54.26 | phigan | yeah? |
| 16:54.31 | phigan | lemme see |
| 16:54.58 | irroot | the h extension ?? is it there does it do something perhaps included from a default |
| 16:55.12 | phigan | i have exten => h,1,Hangup at the end |
| 16:55.22 | irroot | ok |
| 16:55.33 | phigan | lemme try a call to myself and see what console says |
| 16:55.36 | irroot | "set verbose 3" |
| 16:55.38 | irroot | and look |
| 16:58.54 | phigan | hm. |
| 17:00.54 | phigan | I'm guessing it's because I'm doing a bridge.. |
| 17:01.16 | phigan | [2011-05-29 09:59:20] == Starting SIP/1003-08215d20 at default,88005551212,2 failed so falling back to exten 's' |
| 17:01.40 | phigan | [2011-05-29 09:59:20] -- Executing [s@default:1] Answer("SIP/1003-08215d20", "") in new stack |
| 17:01.46 | phigan | that's when the other end hangs up |
| 17:02.17 | irroot | yeah |
| 17:02.47 | irroot | check what dial does when done |
| 17:03.02 | irroot | add a ,2,Hangup() |
| 17:03.14 | irroot | ie after dial looks like a 2 ?? |
| 17:03.37 | phigan | right |
| 17:04.00 | irroot | as it is not hung up it continues |
| 17:04.09 | phigan | well, i have 1,dial and then i,1,hangup .. t,1,hangup .. h,1,hangup |
| 17:04.11 | phigan | ok |
| 17:04.12 | irroot | i play a tone and then hang up |
| 17:04.29 | irroot | 2,hangup is the answer |
| 17:05.22 | phigan | trying it |
| 17:07.17 | phigan | Perfect, that is what I was missing :) |
| 17:07.19 | phigan | Thanks, irroot |
| 17:09.18 | phigan | I had the Hangup in my incoming stuff, but not the outgoing/dialplan |
| 17:09.35 | phigan | exten => s,n,Dial(${PHONE}) exten => s,n,Hangup() |
| 17:10.17 | irroot | pleasure |
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| 18:13.46 | FeyFre | Does any body knows rules of writting topics into asterisk-dev mailing list? Last time(actually first time) I wrote there - got not any reaction. Any ideas? |
| 18:15.02 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 18:16.06 | wdoekes2 | FeyFre: which message? http://lists.digium.com/pipermail/asterisk-dev/ |
| 18:21.12 | FeyFre | wdoekes2: This http://lists.digium.com/pipermail/asterisk-dev/2011-May/049329.html . And damn! I do no know why it become as answer not as new subject. |
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| 18:26.36 | wdoekes2 | because you hit reply instead of creating a new mail |
| 18:29.57 | Freeaqingme | or you happened to have used the same subject and hoaxed the reply-to header |
| 18:30.52 | FeyFre | But I did't(I think) hit reply. Here letter original http://pastebin.com/tkeHLyPF |
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| 18:37.36 | FeyFre | Any way, will it OK if I resubmit my letter again |
| 18:38.39 | wdoekes2 | FeyFre: have you looked at the faxdetect stuff in dsp.c? I'm not so sure there are any hooks you can use |
| 18:43.50 | FeyFre | wdoekes2, a little, but not found any interested hints. I need to catch a bucket of double-toned signals which is present only right after answer. If there is such detector in dsp.c I did not found it. |
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| 18:49.29 | wdoekes2 | well.. if you resubmit your question, you should at least state what you tried and why it's not a good enough solution |
| 18:51.44 | FeyFre | Done it. |
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| 19:49.22 | raden_work | why is skype to asterisk no longer going to be avilable ? |
| 20:04.07 | Freeaqingme | supposedly because skype changed the terms |
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| 20:12.25 | raden_work | skype was bought by M$ |
| 20:12.32 | raden_work | think that has something to do with it ? |
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| 20:58.59 | Hyperbyte | raden_work, it's reasonable to assume that a company would make changes to companies they take over, yes. |
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| 22:35.52 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.4.1 (2011/05/24), 1.6.2.18 (2011/04/26), 1.4.41 (2011/04/26), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
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| 22:41.27 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 23:06.23 | *** join/#asterisk deltaray (~deltaray@unaffiliated/deltaray) |
| 23:06.54 | deltaray | Do both sides of an IAX2 connection need to be the same version? Ie, can I use a 1.4 asterisk server with a 1.6 asterisk server? |
| 23:07.24 | SunTsu | deltaray: Sure, the protocol is the same |
| 23:08.03 | WIMPy | Depending on the new version, you might have to disable call tokens. |
| 23:08.06 | SunTsu | iax is independent of asterisk, heck, you can use all kind of iax software, limiting it to asterisk would be quite stupid |
| 23:09.06 | deltaray | Ok, thanks. |
| 23:10.53 | WiretapWork | SunTsu, IIRC you can't mix <=1.6 & 1.8 IAX |
| 23:11.47 | SunTsu | WiretapWork: er, why not? It's an independent standard |
| 23:12.00 | WIMPy | There have been issues with the first 1.8 versions and 1.6, but it did work with 1.4. |
| 23:12.06 | SunTsu | maybe some details differ, but it should work in general |
| 23:12.43 | WIMPy | IIRC ut was some endiannes bug. |
| 23:13.03 | WIMPy | it |
| 23:16.12 | *** join/#asterisk dachary (~loic@freenode/sponsor/dachary) |
| 23:17.16 | dachary | Hi, I'm trying to make sense of "[May 30 01:14:49] Looking for 0484250905 in from-ovh (domain 10.10.60.4)" followed by "SIP/2.0 404 Not Found" which presumably means 0484250905 was not found |
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| 23:20.37 | dachary | quintana: hi |
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| 23:27.01 | dachary | I'm quite sure I'm not equipped to understand what this error message means. I can't even figure out what to ask :-) from-ovh is an IPBX context which contains 0484250905 in "incoming" calls. I have no clue what "domain" means. The IP is the IP behind the NAT. |
| 23:32.36 | WiretapWork | dachary, you can ignore the domain |
| 23:33.01 | WiretapWork | all it is saying is that it cannot find a match for that number in that context |
| 23:33.18 | WiretapWork | the domain is the other half of the SIP URI |
| 23:33.51 | WiretapWork | 0484250905@10.10.60.4 being the called number if I'm not mistaken |
| 23:42.29 | dachary | WiretapWork: thanks for the hint. I'm trying to make sense of the whole log which I got after sip set debug peer trunk_ovh http://pastebin.com/zXjmM1Ss |
| 23:42.45 | WiretapWork | no problem |
| 23:42.57 | WiretapWork | just make sure your dialplan has a match for that number |
| 23:43.09 | WiretapWork | a SIP 404 is the same as an HTTP 404 in function |
| 23:43.38 | dachary | there is an "Call from '0033485240805' to extension '0485240805' rejected because extension not found." immediately afterwards |
| 23:46.17 | WiretapWork | you haven't got a DID number set up for that extension it would seem |
| 23:46.25 | WiretapWork | are you familiar with your dialplan or are you using freepbx or something? |
| 23:48.17 | dachary | WiretapWork: I'm using http://wiki.xivo.fr/ and I'm not familiar with asterisk indeed. I'm trying to setup an asterisk that receives calls for 1 phone number and does 1 thing : hang up. That seems not too ambitious for a start ;-) |
| 23:48.40 | WiretapWork | pastebin your extensions.conf |
| 23:48.59 | WiretapWork | you also can't HangUp() without Answer() IIRC |
| 23:51.10 | WIMPy | You sure can. |
| 23:51.12 | dachary | http://pastebin.com/LA2PfYbm is extensions.conf but it includes numerous other files |
| 23:51.33 | dachary | it's tricky to debug a configuration created from a web frontend |
| 23:51.54 | WiretapWork | dachary, I was actually hoping to see your entire extensions configuration :P |
| 23:51.58 | WiretapWork | not one file |
| 23:52.35 | dachary | :-) |
| 23:53.00 | dachary | WiretapWork: your hint actually allowed me to resolve the problem. |
| 23:53.07 | WiretapWork | lol |
| 23:53.18 | WiretapWork | SIP debug can seem daunting at times |
| 23:53.25 | WiretapWork | trick is to not letit get to you |
| 23:53.45 | dachary | :-) I went to the "Call management" menu which I assumed to be related to "dialplan" and added an entry for my phone number. It worked. |
| 23:55.50 | WiretapWork | and to think of it like HTTP in a way in the way it works |
| 23:57.25 | dachary | WiretapWork: thanks. I'll get a good night sleep now ;-) |