| 00:06.09 | *** join/#asterisk pentanol (~Unknown@unaffiliated/pentanol) |
| 00:06.22 | pentanol | hi, someone know here cisco voice? |
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| 00:10.31 | [TK]D-Fender | ~polls |
| 00:10.32 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 00:10.57 | [TK]D-Fender | Please ask a more specific and clearer question |
| 00:16.10 | pentanol | someone know translation-rule one of first voice cisco workflo or not? |
| 00:16.32 | pentanol | workflow* |
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| 00:18.23 | pentanol | for instance I have such configuration http://pastebin.ca/2872662 |
| 00:19.25 | pentanol | translation-rule will transfer all calls from 701 through 0/0 and 702 through 0/1 voice ports? |
| 00:19.45 | pentanol | [TK]D-Fender you knows? |
| 00:36.47 | [TK]D-Fender | no |
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| 04:32.19 | ChannelZ | Anyone willing to let me ping flood them for a few seconds? (Not a full-on flood, but every 20ms similar to RTP packets) |
| 04:32.44 | ChannelZ | Trying to track down some maybe network problems |
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| 10:27.30 | stevenm | Hey on SPA devices - is there any good reason to set CPC Duration to 0 instead of the default of 0.5 ? Does CPC Disconnect break some things that don't need it somehow? |
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| 10:53.14 | xochilpili | hi all |
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| 10:54.17 | xochilpili | i have some issues with playback application, i can call to another devices, but i cant hear anything with playback(hello-world), just silence |
| 10:54.20 | xochilpili | any hand? |
| 11:01.34 | xochilpili | please? |
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| 11:11.13 | dkakoti | HI guys how to get the hangup reason if I am doing outbound dial using call file? |
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| 12:27.15 | WIMPy | dkakoti: Use a local channel and the dialplan. Or use AMI instead. |
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| 12:32.18 | marceloamorim | Guys, I`m trying to solve this problem, but I don`t know what to do anymore |
| 12:32.32 | marceloamorim | I have a macro that calls an extension |
| 12:33.22 | marceloamorim | before I call, I set the CALLERID(num) and CALLERID(name), why I use the verbose after I set and didn`t show what I did? =( |
| 12:33.45 | marceloamorim | like, keep showing the ~~s~~ |
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| 12:45.47 | marceloamorim | this is my macro http://pastebin.com/J0MzTBmj |
| 12:48.58 | marceloamorim | the variable ${TRANSFERID} I set at the beggining as global variable -> Set(__TRANSFERID=${CALLERID(num)}); |
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| 13:26.28 | xochilpili | Hello, i have some issues with sounds, playback and voiceemail sounds are not playing |
| 13:26.41 | xochilpili | :( |
| 13:26.44 | xochilpili | any hand? |
| 13:27.29 | xochilpili | i have set on extensions.conf this line: exten=5999,1,VoiceMailMain($(CALLERID(num)),s) |
| 13:29.28 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 13:29.40 | xochilpili | and voiceemail.conf: 5000 => 4711, me, me@email.com |
| 13:30.18 | xochilpili | cli: show <SIP/5000-00000001> Playing 'vm-youhave.ulaw' (language 'en') but no sound here |
| 13:30.58 | WIMPy | Did you answer the call? |
| 13:31.15 | WIMPy | IIRC VoiceMailMain doesn't implicitely do so. |
| 13:32.17 | xochilpili | WIMPy, i call to 5999 (Mailvoice) type user and password for voicemail when ask |
| 13:32.37 | WIMPy | ok |
| 13:33.19 | xochilpili | show messages playing: like " Playing 'vm-youhave.ulaw' (language 'en')" and Playing 'digits/6.ulaw' (language 'en') |
| 13:34.10 | WIMPy | So you get to hear some of the files? |
| 13:34.17 | xochilpili | but theres no sound, also im running asterisk from Vbox, with audio enable, using aplay from shell i can reproduce the files in /var/lib/asterisk/sounds/ |
| 13:34.32 | xochilpili | WIMPy, no i hear nothing... |
| 13:34.41 | xochilpili | WIMPy, that's the issue |
| 13:35.13 | WIMPy | You said you enter user and PIN when asked. So you do that "blind"? |
| 13:35.46 | xochilpili | waching asterisk*cli |
| 13:36.06 | [TK]D-Fender | xochilpili: Show us the call. |
| 13:36.10 | [TK]D-Fender | ~pb |
| 13:36.10 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:36.12 | [TK]D-Fender | ^^^ |
| 13:37.28 | xochilpili | http://pastebin.com/DeMRNaZR |
| 13:38.12 | WIMPy | >>Did you answer the call? - No, you didn't. |
| 13:38.28 | *** join/#asterisk stevenm (~stevenm@stevenm.keele.netcentral.co.uk) |
| 13:38.45 | xochilpili | WIMPy, i call to 5999 |
| 13:38.57 | *** join/#asterisk Tuju (~tuju@214.204.50.195.sta.estpak.ee) |
| 13:39.15 | WIMPy | And that call is never answered. |
| 13:39.20 | xochilpili | WIMPy, sorry, i dont get it |
| 13:39.28 | Tuju | there should be a command: sip show context <c> that would list a combination of dialplan and peers etc |
| 13:39.46 | WIMPy | 'core show application Answer' |
| 13:39.58 | xochilpili | first: i call from zoiper to 5999 the rest you see |
| 13:42.04 | xochilpili | WIMPy, how to answer, then? |
| 13:42.30 | WIMPy | "Answer" |
| 13:42.39 | WIMPy | 'core show application Answer' |
| 13:43.10 | xochilpili | WIMPy, this is what i dont get it, i call to 5999 so, in cli do i have to type: core answer or ..? |
| 13:43.32 | WIMPy | No, you have to add it to your dialplan. |
| 13:44.01 | xochilpili | how is that? |
| 13:44.14 | xochilpili | sorry im really confused, right now! |
| 13:44.37 | [TK]D-Fender | WIMPy: VoiceMail is an implicit answer last I checked... |
| 13:44.44 | WIMPy | You open it with your favourite editor and add that Answer application before calling VoiceMailMain. |
| 13:45.15 | WIMPy | As far as I remember, it isn't. |
| 13:45.17 | [TK]D-Fender | xochilpili: You haven't told us where SIP/5001 is, what kind of device, networking, etc |
| 13:45.36 | [TK]D-Fender | xochilpili: sorry, just saw where you at least said it was Zoiper. |
| 13:45.49 | [TK]D-Fender | xochilpili: Now how is it networked relative to *? |
| 13:47.06 | xochilpili | [TK]D-Fender, yes, i call from zoiper, to 5999, then whaching cli i see playing stuff, that's why im typing user and password for voiceemail whan ask, then i can see, how asterisk play inbox messages, but here i have no sound. |
| 13:47.37 | xochilpili | the same issue with playback(hello-world), cli shows 'playing hello-world' but no sound |
| 13:47.46 | [TK]D-Fender | [08:45][TK]D-Fenderxochilpili: Now how is it networked relative to *? <----------------- |
| 13:48.11 | xochilpili | [TK]D-Fender, networked, what you mean, sorry? |
| 13:48.15 | WIMPy | So we can safely assume that it's zoiper with SIP then. |
| 13:48.28 | [TK]D-Fender | WHAT ROUTERS, SUBNETS, ETC |
| 13:48.34 | [TK]D-Fender | FIREWALLS <- |
| 13:48.39 | WIMPy | You have a network issue. |
| 13:48.54 | [TK]D-Fender | lack of audio being a network issue? Say it ain't so! |
| 13:49.07 | xochilpili | 1.- router (ISP provider), firewall just in centos |
| 13:49.35 | xochilpili | i can make calls, without problems to exten to another... |
| 13:49.37 | WIMPy | And wehere is your Zoiper and where is your Asterisk? |
| 13:49.42 | [TK]D-Fender | xochilpili: Now be VERY clear in your description of where * is running, and the path to where Zoiper is running |
| 13:50.08 | xochilpili | Asterisk is a Centos in Virtualbox, Zoiper is in the host |
| 13:50.17 | xochilpili | is in a** |
| 13:50.54 | [TK]D-Fender | Verify your firewall on CentOS |
| 13:51.19 | [TK]D-Fender | What do you have allowed on it? |
| 13:51.33 | xochilpili | also i deactivate firewall on centos already : same result |
| 13:51.47 | [TK]D-Fender | Show us the call with SIP DEBUG enabled. |
| 13:51.51 | [TK]D-Fender | "sip set debug on" |
| 13:52.17 | xochilpili | http://pastebin.com/w0FDQVvE << iptables centos |
| 13:52.24 | *** join/#asterisk MadHatter42 (~tuwid@109.69.2.62) |
| 13:52.40 | [TK]D-Fender | xochilpili: Trash it |
| 13:52.50 | [TK]D-Fender | -A INPUT -m state --state NEW -m udp -p udp --dport 10000:20000 -j ACCEPT <-- UDP is also stateless..... |
| 13:52.51 | xochilpili | trash what? |
| 13:53.01 | [TK]D-Fender | your entire firewall settings for this test |
| 13:53.03 | xochilpili | remove that line? |
| 13:53.25 | [TK]D-Fender | ALL of them |
| 13:53.30 | xochilpili | ok |
| 13:53.47 | [TK]D-Fender | And dump the actual complete iptables. We need to prove the default action as well |
| 13:54.08 | [TK]D-Fender | For all I know your output is default:deny |
| 13:54.53 | xochilpili | http://pastebin.com/mbCivnaL |
| 13:55.28 | [TK]D-Fender | xochilpili: You just killed the RTP allow |
| 13:55.31 | [TK]D-Fender | alone |
| 13:55.45 | [TK]D-Fender | which means the rest of your policy guarantees failure |
| 13:55.53 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 13:56.05 | xochilpili | shutdown iptables? |
| 13:56.11 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 13:56.12 | [TK]D-Fender | iptables --flush |
| 13:56.14 | [TK]D-Fender | ^^^^^^^ |
| 13:56.24 | [TK]D-Fender | then go get that call with the proper debug |
| 13:56.25 | *** join/#asterisk Milos (~Milos@pdpc/supporter/student/milos) |
| 13:56.48 | xochilpili | http://pastebin.com/3euH3YZq << iptables --flush |
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| 14:00.09 | xochilpili | http://pastebin.com/GQ9NBuDr << debug call to voiceemail |
| 14:01.25 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 14:05.05 | xochilpili | i dont know why is "SIP/2.0 401 Unauthorized" Via: SIP/2.0/UDP 189.251.76.186 << this is my public ipaddr |
| 14:06.20 | [TK]D-Fender | Why are you using your public there? |
| 14:06.32 | [TK]D-Fender | This is a recipe for a HAIRPIN NAT. |
| 14:06.34 | xochilpili | dont know :D |
| 14:06.47 | [TK]D-Fender | Which Stop pointing your devices that way |
| 14:06.49 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-142-164.user.veloxzone.com.br) |
| 14:06.55 | [TK]D-Fender | LOCAL ONLY |
| 14:07.02 | xochilpili | nat=yes? |
| 14:07.09 | [TK]D-Fender | no. |
| 14:07.15 | [TK]D-Fender | stop them from going to that IP in the first place |
| 14:07.20 | [TK]D-Fender | this would be ZOIPER |
| 14:07.26 | xochilpili | i have nat=no already |
| 14:07.36 | [TK]D-Fender | NO |
| 14:07.39 | [TK]D-Fender | FIX ZOIPER |
| 14:08.04 | [TK]D-Fender | <--- SIP read from UDP:192.168.0.3:63542 ---> User-Agent: Z 3.3.25608 r25552 Contact: <sip:5001@189.251.76.186:63542;transport=UDP> |
| 14:08.13 | [TK]D-Fender | Zoiper is looking up your WAN IP and using it in the comms |
| 14:08.29 | [TK]D-Fender | Disable STUN or whatever other NAT handling stuff you have enabled in it |
| 14:08.38 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 14:08.38 | [TK]D-Fender | And make sure you pointed it to the local IP only. |
| 14:09.40 | xochilpili | disabled stun on zoiper and use the local ip |
| 14:10.00 | xochilpili | it works!!! |
| 14:10.01 | xochilpili | :D |
| 14:10.44 | Milos | \o/ |
| 14:11.12 | *** join/#asterisk MKEbrew (4581b2fe@gateway/web/freenode/ip.69.129.178.254) |
| 14:11.37 | Milos | of course it does ;p |
| 14:13.50 | [TK]D-Fender | NEXT!@!@!!! (c) BKW |
| 14:14.05 | xochilpili | lol |
| 14:14.07 | xochilpili | :D |
| 14:14.13 | xochilpili | how can i enable video? |
| 14:14.32 | WIMPy | Off course that wouldn;t have happened with IAX, but I guess noone wants to hear about that. |
| 14:15.09 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:15.09 | [TK]D-Fender | WIMPy: Not sure how routers will handle hairpin on that..... |
| 14:16.01 | WIMPy | Well, it it had faild it wouldn;t have done anything at all, making it rather obvious what's going on. |
| 14:16.15 | [TK]D-Fender | xochilpili: videosupport=yes under [general] , and allow the specific codecs for your peers |
| 14:16.43 | [TK]D-Fender | WIMPy: So like... absolute failure instead of partial ;) |
| 14:16.51 | xochilpili | [TK]D-Fender, wait, i cant access to mailbox :D |
| 14:17.00 | xochilpili | always login incorrect! |
| 14:17.05 | [TK]D-Fender | xochilpili: With dialplan like that... of course not |
| 14:17.21 | [TK]D-Fender | xochilpili: You have not reference your FUNCTION properly |
| 14:17.25 | [TK]D-Fender | +d |
| 14:18.15 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 14:18.40 | [TK]D-Fender | xochilpili: - Executing [5999@paraphones:1] VoiceMailMain("SIP/5001-00000004", "$(CALLERID(num)),s") in new stack <- we couldn't have seen the word "CALLERID" in there like that. It would have evaluated to whatever the callerid on that channel was set to if you had. |
| 14:18.47 | [TK]D-Fender | $( <----------- NO |
| 14:19.26 | *** join/#asterisk Draecos (~Draecos@106-69-27-44.dyn.iinet.net.au) |
| 14:19.27 | xochilpili | [TK]D-Fender, just (CALLERID(num)),s) ? |
| 14:19.35 | [TK]D-Fender | no |
| 14:19.55 | [TK]D-Fender | recheck your instructions on proper syntax for variables & functions |
| 14:20.01 | xochilpili | oks |
| 14:20.42 | WIMPy | And don't forget the mailbox context. |
| 14:21.37 | [TK]D-Fender | Assuming [default] isn't it. |
| 14:21.54 | [TK]D-Fender | (Assuming at all sucks. Always be explicit for things like that |
| 14:22.39 | WIMPy | Don't you have to specify that anyway? |
| 14:22.54 | WIMPy | I seem to remember I fell for that one, too. |
| 14:23.01 | [TK]D-Fender | it will use [default] is you don't specify it |
| 14:23.07 | [TK]D-Fender | if |
| 14:23.12 | [TK]D-Fender | Always has |
| 14:23.34 | [TK]D-Fender | But again, never a good policy to run on assumed working conditions. Own your system and do it right. |
| 14:27.20 | xochilpili | [TK]D-Fender, done! |
| 14:27.33 | xochilpili | now i can login and get the messages |
| 14:27.40 | xochilpili | thank you a lot! |
| 14:27.46 | xochilpili | next challenge; video |
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| 14:59.10 | eppigy | Hello |
| 14:59.14 | eppigy | I am Dave |
| 15:00.34 | WIMPy | I'm afraid I can't do that. |
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| 15:51.21 | Ast001 | Hello what means "Not screen" "Failed screen" and "Passed screen" in SetCallerPres application ? |
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| 15:59.48 | dym | Hey everyone! For some strange Reasons calls to my SNOM 320 are always answered with "forbidden". Does anyone have a clue? Here are sip.conf and cli log: http://pastebin.com/PCRHQTCZ |
| 16:00.23 | [TK]D-Fender | You aren't actually looking at SIP debug |
| 16:01.20 | WIMPy | Ast001: First of all that application is deprecated. Use the CALLERID function. And the SI tell if the ID has been validated. |
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| 16:04.04 | dym | [TK]D-Fender: i did, if you're talking to me. |
| 16:04.35 | WIMPy | dym: Check your security settings and the phones log. |
| 16:05.58 | [TK]D-Fender | dym: I am, and you didn't |
| 16:06.00 | dym | WIMPy: phones log has a lot of provisioning errors - nothing of interest. security settings meaning? sip user types? |
| 16:06.41 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 16:06.47 | WIMPy | The phone has settings equivalent to Asterisks insecure=. |
| 16:06.57 | dym | WIMPy: the phone itself? |
| 16:07.11 | WIMPy | yes |
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| 16:07.24 | dym | [TK]D-Fender: http://pastebin.com/PCRHQTCZ |
| 16:07.31 | Ast001 | WimPy I am trying to achieve to overwrite callerid in google chan_dongle. I am unable to do that with CALLERID(num) and CALLERID(name). I managed to get unknown after setting callingpres=prohib in dongle.conf. I wonder is it possible to replace my dongle's number with other when calling (someone else is calling over my dongle and I wan't his number to be displayed on person my dongle is calling. |
| 16:08.03 | [TK]D-Fender | dym: insecure=invite,port <- you should not have this for phones |
| 16:09.01 | [TK]D-Fender | dym: also that is not proper full debug in-line with teh call |
| 16:09.10 | [TK]D-Fender | dym: Go do another COMPLETE attempt and pastebin it |
| 16:09.27 | dym | [TK]D-Fender: dont be so harsh :( |
| 16:09.36 | dym | it's friday after all! |
| 16:09.49 | WIMPy | Ast001: You can't. I haven't heard of any mobile operator allowing you to do so. I fact the callerID isn't ever sent. |
| 16:10.00 | [TK]D-Fender | Ast001: No way in hell |
| 16:11.23 | dym | [TK]D-Fender: http://pastebin.com/PCRHQTCZ |
| 16:13.16 | [TK]D-Fender | <--- SIP read from UDP:10.1.55.71:2048 ---> SIP/2.0 403 Use Proxy |
| 16:13.18 | [TK]D-Fender | ^ WTF |
| 16:13.33 | marceloamorim | guys, anyone read my problem about callerid? I didn`t got messages after that, if you didn`t and could help me. seems weird |
| 16:14.11 | WIMPy | [TK]D-Fender: I assuem that the allow only calls from configured proxy setting. |
| 16:14.32 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 16:14.35 | WIMPy | Whatever the name of that parameter is. |
| 16:14.37 | [TK]D-Fender | marceloamorim: I see no question from you in the past 3 1/2 hours |
| 16:14.50 | dym | [TK]D-Fender: im kinda puzzled too. |
| 16:15.04 | [TK]D-Fender | dym: Go remove any proxy settings from your phone |
| 16:15.11 | [TK]D-Fender | dym: It should be direct |
| 16:15.35 | dym | [TK]D-Fender: here comes the funny thing: i did not set any :D |
| 16:15.39 | dym | ill have a look though |
| 16:16.37 | dym | http://drop.openroot.de/14A1K/2TPPR8qI |
| 16:16.42 | WIMPy | marceloamorim: That global bit puzzels me. Are you trying to use that in a dynamic way? That's not how it works. Globals are evaluated once when the dialplan is read. |
| 16:16.48 | dym | theres nothing that qualifies a "proxy setting" |
| 16:17.39 | marceloamorim | well, the global variable seems works pretty well, because when I verbose the ${TRANSFERID} shows me the right thing, |
| 16:18.36 | Ast001 | Ok thanks then I will send unknown. |
| 16:18.49 | marceloamorim | [TK]D-Fender: I ask about the callerid, because I`m setting the callerid before I call, and I didn`t get the same information =( |
| 16:18.57 | *** part/#asterisk Ast001 (~uros@82.117.198.142) |
| 16:18.58 | marceloamorim | I`ll paste the context |
| 16:19.16 | [TK]D-Fender | dym: Check other more general SIP settings as well |
| 16:19.54 | marceloamorim | http://pastebin.com/ky9wkU8q |
| 16:20.34 | marceloamorim | I even get the same callerid after I set the callerid and the next line I verbose the variable =( |
| 16:21.29 | dym | [TK]D-Fender: This is the settings of the target, or source phone? |
| 16:22.15 | [TK]D-Fender | marceloamorim: we have no proof of what your call looks like, if it is using that code at all, it ofther things are set in the call flow etc. |
| 16:22.22 | [TK]D-Fender | marIn effect that shows us NOTHING |
| 16:22.30 | [TK]D-Fender | marceloamorim: Show us the call |
| 16:22.48 | [TK]D-Fender | dym: in that other tab |
| 16:23.00 | [TK]D-Fender | dynYou're in "login", and I see a "SIP" tabe |
| 16:23.19 | [TK]D-Fender | dym: and 2at least 1 other that I'm automatically suspicious of |
| 16:23.32 | WIMPy | dym: Advanced settings -> QoS/Security -> Filter Packets frrom Registrar |
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| 16:29.13 | dym | WIMPy: done. still same |
| 16:31.09 | WIMPy | Hmm. And nothing usefull in the log, even if you increase to max? |
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| 16:39.37 | marceloamorim | [TK]D-Fender: I know that you don`t like pastebins like that. but I paste when the callerid was set and didn`t show the right thing. but don`t be mad with me plz =) http://pastebin.com/KGWepviP |
| 16:42.09 | WIMPy | marceloamorim: We don't see you setting anything there. |
| 16:43.15 | [TK]D-Fender | marceloamorim: |
| 16:43.23 | [TK]D-Fender | Set(${CALLERID(num)}=${TRANSFERID}); |
| 16:43.24 | [TK]D-Fender | Set(${CALLERID(name)}=${TRANSFERID}); |
| 16:43.28 | [TK]D-Fender | that is NOT how you set functions. |
| 16:44.21 | [TK]D-Fender | The sort omistake you should have noticed the moment you see output like this : -- Executing [~~s~~@dial_sip_transfer:13] Set("Local/602@ramais_transfer-00000417;2", "1415982854=201411141934") in new stack |
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| 16:53.15 | dym | WIMPy: could have been the case that your setting was "the one". |
| 16:53.18 | dym | now it works. |
| 16:56.59 | marceloamorim | I learn at voip-info |
| 16:57.34 | [TK]D-Fender | marceloamorim: You have failed to learn the very basics of using functions. |
| 16:57.43 | [TK]D-Fender | marceloamorim: And you should be using that WIKI as a LAST resort. |
| 16:57.48 | [TK]D-Fender | marceloamorim: Not first |
| 16:58.06 | marceloamorim | =( |
| 17:01.02 | anonymouz666 | marceloamorim: stop crying. use something like this: Set(CALLERID(name)=${CALLERID(name)}) |
| 17:02.41 | anonymouz666 | read the basics |
| 17:02.52 | marceloamorim | anonymouz666: I`m not crying!! I just didn`t know what was wrong, I was telling that I learn at voip-info when the subject was callerid |
| 17:03.14 | anonymouz666 | voip-info is the best place to read old stuff |
| 17:03.20 | marceloamorim | I already use the same code to change the callerid |
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| 17:19.39 | [TK]D-Fender | marceloamorim: Fix your code. |
| 17:20.16 | anonymouz666 | the best quote ever about fix is... "I broke it, please fix" |
| 17:20.32 | anonymouz666 | hehe |
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| 17:29.52 | WIMPy | dym: Then you should find out why you need that setting. |
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| 17:31.54 | marceloamorim | I didn`t broke brother, I developer my system, the problem is, I set the same code in other context and I didn`t have a problem, but thx for point me that the code is wrong |
| 17:32.47 | [TK]D-Fender | Same code in another context will fail just the same |
| 17:33.02 | [TK]D-Fender | A different context will not make that magically work |
| 17:33.30 | marceloamorim | I`m using the AEL, and the AEL so far was set doing the same |
| 17:34.22 | [TK]D-Fender | It sure wouldn't work elsewhere |
| 17:35.00 | marceloamorim | my variable ${TRANSFERID} got the right number, I just can`t transfer the ${TRANSFERID} to CALLERID, I`ll figure out |
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| 17:37.35 | [TK]D-Fender | marceloamorim: I'm not sure what part of "your Set is wrong because you aren't using functions properly" you are having diffulty with. |
| 17:38.11 | marceloamorim | anonymouz666: already give me motivation, don`t worry [TK]D-Fender, I`ll figure it out\ |
| 17:39.39 | anonymouz666 | sorry I don't use AEL, can't help |
| 17:39.48 | [TK]D-Fender | Isn't even an AEL issue |
| 17:39.55 | [TK]D-Fender | the syntax for this is the same |
| 17:40.04 | anonymouz666 | is it? |
| 17:40.05 | anonymouz666 | ok |
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| 17:59.37 | Stefan27 | if i have two simple bridges, [SIP_A,Local1] , [SIP_B,Local,2] (A talking to B) and A tries to transfer B by sending a SIP msg with 'REFER-To C'; then what variable governs the From:-Header in the invite sent to C? because in my weird test case the invite to C has a from field with A not B as expected |
| 18:00.59 | WIMPy | Attended transfer or blind transfer? |
| 18:01.05 | Stefan27 | blind |
| 18:01.09 | WIMPy | Or that "blonde" transfer? |
| 18:02.07 | anonymouz666 | hah |
| 18:02.21 | Stefan27 | i dumped the chanvars of SIP_A and Local1 before doing the transfer |
| 18:03.46 | Stefan27 | does the output of "dialplan show chanvar Local1" determine the From-header for invites to C, if SIP_A (Local1's bridgepeer) should refer him to C? |
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| 18:07.32 | Stefan27 | because the only variable in Local1 that i can see that relates to B is 'CALLED_NUMBER', all other chanvariables of Local1 relates to A |
| 18:09.31 | Stefan27 | the test case setup before the transfer is a manager-originated call from asterisk (call that channel Local1) to SIP_A, which upon answering calls SIP_B, creating the two bridged i meentioned |
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| 18:46.20 | ChannelZ-Wk | So I'm trying to evaluate SIP call quality and have been doing some ChanSpy()ing on calls to at least listen to the incoming for dropouts, etc. It seems like when I do the initial ChanSpy, asterisk suddenly reports a TON of RX packet loss in a burst for the 'local' SIP channel leg of the call. |
| 18:49.27 | ChannelZ-Wk | IE: Attaching SIP/Bob-00000040 to SIP/Vitelity-Out-00000042 -- I am SIP/Bob, SIP/Vitelity-Out is the external call. When I immediately look at channelstats though, I see 5203 packets lost on SIP/Confroom which is the channel bridged to the SIP/Vitelity-Out call I'm listening to. |
| 18:52.13 | ChannelZ-Wk | This has to just be an anamolous reading because that many lost packets would be like a substantial amount of time (20ms a packet?) |
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| 21:17.35 | xochilpili | hi all |
| 21:18.10 | xochilpili | i have some issues trying to configure a tronk |
| 21:18.14 | xochilpili | any hand? |
| 21:18.23 | [TK]D-Fender | Maybe if you show us the problem |
| 21:18.35 | [TK]D-Fender | Show first, response second. |
| 21:18.37 | WIMPy | Or tell us what kind of trunk first. |
| 21:18.42 | xochilpili | [TK]D-Fender, hi again! |
| 21:19.33 | xochilpili | well, i got 'Connection errro 63' from zoiper |
| 21:20.12 | xochilpili | i register from sip.conf, but im a little bit confused (as you know) |
| 21:20.27 | xochilpili | i add a register in sip.conf*** |
| 21:20.42 | [TK]D-Fender | xochilpili: To be practical your first thought coming in should be something like : "I'm trying to accomplish X, I created Y (pastebin, etc) to try to do it, I get bad result Z (Pastebin, etc). Can someone help identify the source of the problem?" |
| 21:21.09 | [TK]D-Fender | [16:19]xochilpiliwell, i got 'Connection errro 63' from zoiper <-so far that doesn't sound like a problem with a TRUNK. |
| 21:21.27 | [TK]D-Fender | xochilpili: That sounds like a probalme with just your server & a local client |
| 21:21.38 | [TK]D-Fender | xochilpili: not some telco service provider |
| 21:22.48 | xochilpili | http://pastebin.com/CKaGwW4a |
| 21:22.56 | xochilpili | there's my configuration |
| 21:23.01 | [TK]D-Fender | and the debug? |
| 21:23.12 | xochilpili | pasting... |
| 21:24.03 | [TK]D-Fender | xochilpili: [cmas-out] <- this doe not have a defined context to send calls to. And your dilplan makes no use of it to call OUT. |
| 21:24.23 | [TK]D-Fender | xochilpili: You didn't configure this to be used at all |
| 21:24.28 | xochilpili | http://pastebin.com/Ud6itjVR < debug |
| 21:24.36 | [TK]D-Fender | xochilpili: nothing you dial on Zoiper leads to using it. |
| 21:24.58 | xochilpili | thats the issue :D im realy confused |
| 21:25.35 | xochilpili | i change: exten => _9.,1,Dial(SIP/$EXTEN@cmas-out,30,Ttm) |
| 21:25.38 | [TK]D-Fender | <--- SIP read from UDP:192.168.0.77:42313 ---> REGISTER sip:asterisk.paranoids.us:5060;transport=UDP SIP/2.0 To: <sip:5002@asterisk.paranoids.us:5060;transport=UDP> |
| 21:25.52 | [TK]D-Fender | Why are you using a FQND there? |
| 21:25.54 | [TK]D-Fender | You are LOCAL |
| 21:26.13 | WIMPy | And again a syntax error. |
| 21:26.35 | WIMPy | Read the dialplan basics in the |
| 21:26.37 | WIMPy | ~book |
| 21:26.37 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:26.44 | WIMPy | You will need them a lot more. |
| 21:29.30 | xochilpili | [TK]D-Fender, should be on extensions.confg [cmas-out] right? |
| 21:30.10 | [TK]D-Fender | xochilpili: it should be wherever you want calls coming FROM them to land in. |
| 21:30.50 | ChannelZ-Wk | MY TRONK IS BROKEN |
| 21:30.55 | [TK]D-Fender | packs up to head home |
| 21:30.58 | [TK]D-Fender | BBIAB |
| 21:34.06 | WIMPy | I have just released a new version of tracI, if anyone here cares. http://www.yeti.dk/lsoft/tracI-0.22.tar.bz2 |
| 22:02.14 | *** join/#asterisk krapper (krapper@gibson.krapper.net) |
| 22:16.52 | *** join/#asterisk toresbe (~toresbe@158.36.190.254) |
| 22:17.33 | toresbe | Heya! I've been trying to Google this to no avail. I have a seriously weird thing I want to try. I'm about to get a 1991-vintage video telephone, which I'm just dying to try out. |
| 22:17.35 | *** join/#asterisk tzafrir (~tzafrir@bzq-84-109-18-138.red.bezeqint.net) |
| 22:18.15 | toresbe | It's ISDN-based. Can asterisk work with video telephony over an ISDN channel? |
| 22:18.27 | WIMPy | No |
| 22:19.06 | WIMPy | It cannot even support other voice CODECs. |
| 22:19.13 | WIMPy | .. than G.711. |
| 22:20.29 | toresbe | Dang. |
| 22:21.27 | toresbe | I suppose it would be a rather substantial undertaking to provide the necessary support. |
| 22:22.10 | WIMPy | I don't think so. Not if you have another video phone that uses the same video CODEC that is. |
| 22:22.30 | WIMPy | There's no video transcoding. |
| 22:24.10 | toresbe | I don't even know which codec this thing uses; it's 1991-vintage. |
| 22:25.08 | toresbe | I can't find any information on it at all. It's a Tandberg Vision - a friend works at the part of Cisco which used to be Tandberg, and he said he'd look for some docs. |
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| 22:27.04 | toresbe | I think the audio component actually might be G.711. |
| 22:27.35 | WIMPy | Does it use one or two channels? Or even more? |
| 22:28.08 | toresbe | I'm not sure. I don't even have it here yet. But I assume it uses two, at least. |
| 22:28.13 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:28.18 | WIMPy | But audio codec is less of an issue. |
| 22:30.10 | joren | has anyone had any success with the gtalk pluggin these days? It seems like incoming calls are showing up but when I try to call out I get "gtalk_request: Could not find recipient." |
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| 22:55.27 | joren | weird, I got one call to pass through but now it's back to that same error |
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| 22:58.00 | *** join/#asterisk blistov (~bens@S0106e03f490494e9.vs.shawcable.net) |
| 22:58.18 | blistov | Hey all. I've got my Asterisk server sitting on my LAN, behind a firewall with two WAN's. |
| 22:58.29 | blistov | Wan1 is primary, Wan2 is only used as a failover. |
| 22:59.22 | blistov | They both have different IP's. The SIP trunk is setup to automatically fail over from one IP to the other, but the RTP isn't getting the hang up because Asterisk's "external IP" is set to the WAN ip of WAN1. |
| 22:59.27 | blistov | How do I get around this? |
| 22:59.35 | blistov | I suspect I'm missing something pretty basic here. |
| 23:00.19 | blistov | Also, it seems silly to me that my Asterisk server is WAITING for the SIP trunk provider to decide that WAN1 is down, and to then initiate a new session on WAN2, instead of Asterisk just calling the trunk provider from the new IP. |
| 23:00.26 | blistov | WTF am I missing here? |
| 23:04.38 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 23:08.33 | [TK]D-Fender | [17:59]blistovHow do I get around this? <- write your own script to poll for the IP changeover and update the value and apply it |
| 23:08.48 | [TK]D-Fender | Or use a dyndns service on a suitable interval. |
| 23:08.51 | [TK]D-Fender | etc |
| 23:09.11 | [TK]D-Fender | You aren't missing anything. The solution is pretty much on you. It isn't a "feature" per-se |
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