| 00:00.56 | *** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony) |
| 00:02.13 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-196-000.sc.res.rr.com) |
| 00:02.31 | ChrisInSydney | Still no one else in the bridge |
| 00:02.49 | WIMPy | Under the bridge? |
| 00:03.21 | ChrisInSydney | that time of the night I guess |
| 00:03.35 | ChrisInSydney | WIMPy: exten => 882,1,Dial(SIP/200901@login.zipdx.com) |
| 00:03.40 | ChrisInSydney | VUC bridge |
| 00:04.05 | WIMPy | hates VOIP |
| 00:16.54 | hardwire | heh |
| 00:27.30 | *** join/#asterisk adeel (~adeel@184.175.36.92) |
| 00:29.22 | ChrisInSydney | adeel: VUC bridge is still up |
| 00:29.35 | ChrisInSydney | adeel: exten => 882,1,Dial(SIP/200901@login.zipdx.com |
| 00:35.25 | p3nguin | chrisinsydney: CNAME www.dhs.gov |
| 00:35.34 | p3nguin | You think that's suitable? |
| 00:35.42 | ChrisInSydney | he he he he |
| 00:35.56 | ChrisInSydney | classic :D |
| 00:36.32 | p3nguin | I was trying to find something of theirs that was a little more mission critical. |
| 00:37.46 | p3nguin | I'm still on the conf, but working on finishing supper now. |
| 00:38.34 | ChrisInSydney | I'm just playing music at it |
| 00:38.48 | ChrisInSydney | still hand editiing the database |
| 00:52.49 | dym | When using call screening, there is a DTFM option (3) to send the caller to the "torture" menu - but when this is called the call is simply terminated. how can i interfere with this DTFM press and send the caller to the context of my choice? |
| 00:53.49 | p3nguin | Pressing 3 ends the Dial() with a DIALSTATUS of TORTURE. |
| 00:54.49 | p3nguin | So you'd do something like Goto(${DIALSTATUS}) and make sure you have a priority of TORTURE in the current extension. |
| 00:55.38 | dym | actually ends like this: |
| 00:55.38 | dym | Oct 15 02:54:49] WARNING[20924]: pbx.c:4088 pbx_extension_helper: No application '1,Goto' for extension (incoming, 4954XXXXXXXX, 7) |
| 00:55.57 | ChrisInSydney | dym, p3nguin, what is needed is to send the person into a conference bridge with bad music and a recording of someone saying" Who were you after again?...hang on I'll seei fI can get them for you"..back to hold music |
| 00:56.09 | ChrisInSydney | When you get two calls thransferred, they can talk to eachother |
| 00:56.38 | p3nguin | I guess you don't have an application by the name of '1,Goto' like it says. |
| 00:56.51 | p3nguin | I.e. your dialplan is broken/wrong. |
| 00:57.11 | ChrisInSydney | dym: or needs some more attention |
| 00:57.23 | p3nguin | Pastebin your dial plan. |
| 00:58.27 | dym | p3nguin: Is this ment to be priority 7 on the number? |
| 00:58.32 | dym | in context incoming? |
| 00:58.54 | ChrisInSydney | maybe priority h for hangup ?? |
| 00:59.04 | dym | ChrisInSydney: no. |
| 00:59.05 | p3nguin | I'm not sure. Just show me your dial plan and I'll tell you what's wrong. |
| 00:59.19 | p3nguin | There's no priority h that I know of. |
| 00:59.21 | ChrisInSydney | I'm just guessing as I haven't fone this before |
| 00:59.22 | dym | sec - ill try fixing it myself |
| 00:59.47 | ChrisInSydney | so what does it do in the CLI, just hang up ?? |
| 01:00.02 | p3nguin | I already addressed that issue. |
| 01:00.41 | dym | KACHING |
| 01:00.43 | dym | fixed |
| 01:00.43 | dym | :D |
| 01:00.49 | dym | it was actually prio 7 in the context |
| 01:01.01 | dym | so i could kick it into the torture menu by Goto'ing |
| 01:01.03 | ChrisInSydney | sorry, h extension, not priority |
| 01:01.07 | ChrisInSydney | my bad |
| 01:01.15 | dym | thanks anyways p3nguin |
| 01:01.26 | dym | more fun fixing it myself than beeing spoonfed :P |
| 01:01.33 | ChrisInSydney | dym: Cool. SOmetimes you just have to type it to someone else and it makes sense |
| 01:01.39 | ChrisInSydney | all of a sudden |
| 01:01.59 | dym | ChrisInSydney: well - the output kinda said it all |
| 01:02.03 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 01:02.05 | ChrisInSydney | Thats OK I'm happy to take the credit though :D |
| 01:02.17 | dym | for what? |
| 01:02.55 | dym | Brilliant :D Torture menu with voice prompts recorded by myself. Let total confusion commence! :D |
| 01:02.56 | ChrisInSydney | never mind |
| 01:03.26 | ChrisInSydney | dym: why dont you jump on to the VUC conference bridge |
| 01:03.36 | ChrisInSydney | exten => 882,1,Dial(SIP/200901@login.zipdx.com) |
| 01:03.42 | ChrisInSydney | g711 / g722 |
| 01:03.50 | dym | what would i do there? |
| 01:03.59 | ChrisInSydney | p3nguin and I are still on |
| 01:04.02 | dym | oh |
| 01:04.14 | ChrisInSydney | you can talk |
| 01:04.24 | ChrisInSydney | and we (i ?) will talk back |
| 01:04.36 | *** join/#asterisk SwK (~SwK@freeswitch/developer/swk) |
| 01:04.57 | ChrisInSydney | there is a confernce cal every Friday at 12:00pm eastern ?? check out http://vuc.me |
| 01:05.35 | WIMPy | vuc.me? do you get assimilated? |
| 01:06.03 | dym | seems so |
| 01:06.05 | p3nguin | DIALSTATUS priorities - http://pastebin.com/FEXndLUc |
| 01:09.28 | p3nguin | I think it's <your PIN>*929 |
| 01:09.37 | p3nguin | But when I call it, it never works for me. |
| 01:10.03 | p3nguin | 20:10 |
| 01:10.46 | p3nguin | Am I muted? |
| 01:16.41 | PhoenixMage | back |
| 01:16.53 | *** join/#asterisk jetlag (jetlag@pool-71-168-246-8.cmdnnj.east.verizon.net) |
| 01:17.01 | p3nguin | HTTP request failed: 404 File Not Found |
| 01:17.03 | dym | <-- Patrick btw. |
| 01:17.17 | p3nguin | Stream is down. |
| 01:18.46 | *** join/#asterisk jblack (~jblack@75-149-160-4-Washington.hfc.comcastbusiness.net) |
| 01:19.31 | ChrisInSydney | jetlag: VUC bridge is still up |
| 01:19.41 | ChrisInSydney | exten => 882,1,Dial(SIP/200901@login.zipdx.com) |
| 01:19.46 | ChrisInSydney | come join us |
| 01:20.18 | p3nguin | He almost sounded like it used a lot of resources to have it open. |
| 01:20.27 | p3nguin | Griping about your typing. |
| 01:20.48 | p3nguin | I don't even know why it would have needed to be recorded. |
| 01:25.21 | PhoenixMage | anyone got an example dialplan.xml they can post? |
| 01:26.49 | WIMPy | xml? |
| 01:26.56 | WIMPy | Is this some gui stuff? |
| 01:27.07 | p3nguin | Cisco phone file |
| 01:27.18 | WIMPy | Oh |
| 01:27.49 | WIMPy | Leave it empty? That' been the only sensible configuration I found so far for "dialplans". |
| 01:28.19 | p3nguin | It just takes a while for the dialed number to actually send if you don't have one. |
| 01:28.25 | PhoenixMage | p3nguin: You said my prob wouldnt be related to it, any further theories on why I cant dial? |
| 01:28.35 | p3nguin | I couldn't think of anything. |
| 01:29.14 | p3nguin | If you dial with the handset on-hook, then press Dial, and you still experience the problem, I don't know what else there is. |
| 01:30.03 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 01:30.03 | WIMPy | Yes, you have to press send unless you can use overlap sending, but there's nothing to do about that anyway. |
| 01:32.17 | dym | wtf |
| 01:32.29 | dym | Reminded me of some Asterisk preset soundfile |
| 01:32.31 | WIMPy | loca people |
| 01:34.57 | dym | <-- just testing something i need the phone for. |
| 01:40.17 | dym | p3nguin: i wonder why this recording never actually reaches me: http://pastebin.com/3L3yLvJc |
| 01:40.27 | dym | mail is send, as seen from mail.log too |
| 01:42.40 | p3nguin | I'm not even sure what you're trying to do. |
| 01:42.46 | PhoenixMage | brb |
| 01:42.46 | *** part/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 01:42.55 | ChrisInSydney | <dym> Not that many people find their way there though, unfortunately. |
| 01:42.55 | ChrisInSydney | <dym> I wondered why there aint no permanent conference bridge for asterisk interested people. |
| 01:42.55 | ChrisInSydney | <dym> Maybe something like that should be instated. |
| 01:43.35 | *** join/#asterisk Beave (~champ@bundy.vistech.net) |
| 01:43.39 | p3nguin | I had thought about creating a somewhat permanent conf for #asterisk people. |
| 01:43.58 | ChrisInSydney | sip:asterisk@somebodysdomain.com |
| 01:43.59 | p3nguin | I don't know if it would be used or not. |
| 01:44.07 | ChrisInSydney | it wouldbnt matetr |
| 01:44.09 | ChrisInSydney | matter |
| 01:44.18 | dym | p3nguin: i also have the resources. |
| 01:44.24 | dym | ill see to it |
| 01:44.27 | p3nguin | If I'm going to bother setting it up, I'd like for it to be used. |
| 01:44.47 | dym | I dont care if its used (: At least then there is some sort of "entry point" |
| 01:44.59 | Beave | p3nguin: tht's sorta what telephreak.org did.. for years now. |
| 01:45.13 | p3nguin | There's one already running? |
| 01:45.18 | Beave | and probably countless other people. |
| 01:45.22 | dym | Beave: point us towards it |
| 01:45.25 | Beave | p3nguin: just voip hobbyist stuff. |
| 01:45.29 | Beave | http://www.telephreak.org |
| 01:45.45 | *** join/#asterisk gogasca (~Adium@nat/cisco/x-srqurpqmegapuaer) |
| 01:46.23 | dym | Dont see info to a permanent conference there tho |
| 01:46.39 | Beave | option # 2 in the conf. "general conf". |
| 01:46.43 | gogasca | hi ppl anyone has tested the *10 video bridge? im wondering if I place a conf call using 3 tandberg h264 720p quality will be preserved? |
| 01:46.45 | dym | http://www.telephreak.org/?pbx_voip |
| 01:46.56 | Beave | not used as much as it use to be, however. |
| 01:47.25 | dym | Well lets see how things deliver. Would be cool to have something dedicated with multiple bridges - language wise |
| 01:48.09 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 01:53.44 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:55.12 | p3nguin | Anyone know if there is a way to make a Cisco 7960G ring on a second call rather than give a call-waiting tone on the speaker? |
| 01:56.05 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 01:57.21 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 01:57.23 | gogasca | i think i know how |
| 01:57.34 | gogasca | gimme a sec |
| 01:57.41 | gogasca | let me open up my config |
| 01:58.21 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 02:00.16 | p3nguin | <PROTECTED> |
| 02:03.35 | *** join/#asterisk jetlag (jetlag@pool-71-168-246-8.cmdnnj.east.verizon.net) |
| 02:15.36 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:16.53 | p3nguin | gogasca: Did you find anything useful? |
| 02:20.22 | *** join/#asterisk snowbot1 (~lmr@ppp118-208-159-167.lns20.bne1.internode.on.net) |
| 02:20.28 | snowbot1 | hi all |
| 02:21.14 | gogasca | oh yeah |
| 02:21.37 | gogasca | in cucm there is a setting called: Ring Setting (Phone Active) |
| 02:21.47 | gogasca | just need to find what is the xml under that line |
| 02:22.27 | gogasca | the options we have is:flash, ring once, beep only |
| 02:22.33 | gogasca | system default |
| 02:22.45 | gogasca | so take a look at your xml and see if u find something similar |
| 02:23.29 | gogasca | that setting is at line level not device level |
| 02:23.50 | gogasca | sorry man im in middle of some upgrade |
| 02:24.23 | gogasca | u need to make sure ur Maximum Number of Calls is > 2 and busy trigger > 2 as well |
| 02:26.03 | snowbot1 | is anyone free to ask questions here? :) |
| 02:26.49 | p3nguin | ~ask |
| 02:26.50 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:27.28 | p3nguin | dijib? |
| 02:27.50 | WIMPy | I think you wanted to say "yes". |
| 02:28.30 | ChrisInSydney | Hey all: exten => 882,1,Dial(SIP/200901@login.zipdx.com) |
| 02:28.43 | ChrisInSydney | The VUC conference bridge is still up !!! |
| 02:28.59 | snowbot1 | thanks |
| 02:41.51 | PhoenixMage | I could have sworn that 7975 used to pull firmware from a subdir but now it wonts everything for a firmware upgrade in the tftp root |
| 02:42.22 | p3nguin | gogasca: I can't seem to find much information about adding that setting to my asterisk system. |
| 02:42.43 | gogasca | that will be in your 7960 xml config file |
| 02:43.45 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 02:43.53 | p3nguin | My XMLDefault.cnf.xml is pretty minimal, and my SEP<MAC>.cnf.xml is even smaller. I guess I need a full list of all options. |
| 02:55.30 | snowbot1 | wooo i got my first asterisk setup working :) iax2 trunked |
| 03:00.55 | *** join/#asterisk mintos (~mvaliyav@114.143.165.134) |
| 03:08.23 | *** join/#asterisk radic (~radic@dslb-094-216-242-102.pools.arcor-ip.net) |
| 03:16.08 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 03:20.38 | dijib | i like it better |
| 03:21.18 | p3nguin | I found the ringSettingActive parameter. |
| 03:21.40 | p3nguin | Maybe that's the one I need. |
| 03:28.20 | dijib | crazy world i still dont think that you. |
| 03:28.31 | p3nguin | *shrug* |
| 03:29.08 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 03:34.47 | *** join/#asterisk tmrhmdv (~tmrhmdv@ool-4575afcd.dyn.optonline.net) |
| 03:37.00 | tmrhmdv | Hi, folks! I'm trying to setup Asterisk and Ekiga (softphone), but getting "Could not register (Remote host party is offline)", but they are online, here's the info: http://pastebin.com/4jYrc2WG what could it be from? |
| 03:39.13 | PhoenixMage | p3nguin: Seems it was a dialplan.xml problem |
| 03:39.30 | p3nguin | So you had a dialplan.xml that was broken? |
| 03:40.14 | PhoenixMage | no I had no dialplan.xml at all |
| 03:40.18 | WIMPy | tmrhmdv: Why do you use AMI instead of the *CLI? |
| 03:40.23 | PhoenixMage | then I had an empty one |
| 03:40.36 | PhoenixMage | now I have one I picked up from voip-info |
| 03:41.32 | tmrhmdv | WIMPy: I had actually found an easier software for Asterisk called "A2Billing" and followed a guide. But when I reach the Softphone part, it's giving me those errors |
| 03:46.41 | WIMPy | tmrhmdv: If your phone tells you the server is offline, that must be a networking issue. |
| 03:47.07 | tmrhmdv | WIMPy: From my end or the server? |
| 03:48.53 | dijib | call waiting behaviour |
| 03:49.58 | tmrhmdv | What should the "bindaddr" be set to in manager.conf? my server IP or 127.0.0.1? |
| 03:50.05 | tmrhmdv | Currently it is 127.0.0.1 |
| 03:50.30 | p3nguin | level 1: start=2011-10-14 10:50:03 |
| 03:50.41 | p3nguin | <PROTECTED> |
| 03:52.12 | p3nguin | level 1: billsec=43323 |
| 03:52.18 | p3nguin | :/ |
| 03:53.09 | p3nguin | tmrhmdv: If you want other computers on the network to have access to it over IP, you'll have to set it to the address on the ethernet interface. |
| 03:53.54 | dijib | kylie menode? |
| 03:54.26 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 03:55.01 | p3nguin | en.wikipedia.org/wiki/Kylie_Minogue |
| 03:55.43 | dijib | and the wiki for who sponseres this cof |
| 03:55.44 | dijib | ? |
| 03:55.46 | dijib | bridge |
| 03:55.49 | dijib | conference |
| 03:56.15 | ChrisInSydney | he he |
| 03:56.22 | ChrisInSydney | I'm outa here |
| 03:56.26 | ChrisInSydney | off to a wedding |
| 03:56.36 | dijib | type type i hear |
| 03:56.41 | ChrisInSydney | get some more people into the bridge so its up whe i get back |
| 03:56.46 | p3nguin | http://www.zipdx.com/ |
| 03:57.05 | ChrisInSydney | Everybody: exten => 882,1,Dial(SIP/200901@login.zipdx.com) |
| 03:57.07 | p3nguin | I'll be gone soon, too. |
| 03:57.13 | ChrisInSydney | all cool |
| 03:57.20 | ChrisInSydney | <PROTECTED> |
| 03:57.23 | dijib | if i was ever allowed back in /g/tech...... |
| 03:57.30 | dijib | bye chris thnx for the invite |
| 03:58.30 | p3nguin | 882 is VUC on teh keypad. |
| 04:12.37 | dijib | 88351000090146 |
| 04:13.27 | dijib | p3nguin, are you using iNum or Virtual numbers? |
| 04:13.38 | dijib | and did that conf shut down with music? |
| 04:14.50 | p3nguin | If you heard music, that was probably mine. |
| 04:15.02 | dijib | oh sounded like hold music |
| 04:15.06 | p3nguin | It was. |
| 04:15.20 | dijib | too fluide to be your during the dos no? |
| 04:15.23 | p3nguin | See if you hear it again. |
| 04:15.31 | dijib | i hung up |
| 04:15.34 | p3nguin | oh |
| 04:15.36 | dijib | wife was about to kick my ass |
| 04:15.53 | p3nguin | It says two other participants, so I figured it was the Chrises. |
| 04:16.23 | p3nguin | iNum or virtual numbers... for what? |
| 04:18.58 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 04:20.18 | *** join/#asterisk EugeneKay (eugene@itvends.com) |
| 04:21.40 | EugeneKay | Hi, all. Looking into setting up an Asterisk server for my small business. Looking for a recommendation on a SIP provider who I can get a 1-800(or any of the other toll-free prefixes) from. |
| 04:21.47 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 04:22.41 | p3nguin | ~itsp |
| 04:22.41 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 04:23.04 | *** join/#asterisk mintos (~mvaliyav@114.143.165.134) |
| 04:26.08 | *** join/#asterisk dym (~patrick@unaffiliated/dym) |
| 04:26.11 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 04:26.20 | EugeneKay | spams the bot in private |
| 04:37.14 | p3nguin | There it went. |
| 04:41.47 | dym | There it went! |
| 04:42.50 | p3nguin | Were you still on? |
| 04:42.58 | dym | On what? |
| 04:43.03 | dym | The conference? |
| 04:43.07 | p3nguin | Yes. |
| 04:43.13 | dym | Nah, left a couple of hours ago |
| 04:43.52 | p3nguin | I left it connected, but I guess I needed to make noise to keep it going. |
| 04:46.14 | dijib | EugeneKay, voip.ms |
| 04:47.24 | dijib | did it die? |
| 04:47.39 | EugeneKay | They treat you good? |
| 04:47.52 | dijib | yes fair from what ive seen |
| 04:48.01 | dijib | immediate attention when you open a ticket |
| 04:48.04 | dijib | within 6hours |
| 04:51.01 | SeRi | ~itsplist-us |
| 04:51.01 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 04:59.50 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:03.39 | p3nguin | seri: So what do you recommend for this phone? Hunt for an appropriate SIM? |
| 05:07.46 | SeRi | p3nguin, That would be hard. |
| 05:07.52 | SeRi | well not impossible |
| 05:08.03 | SeRi | but very specific |
| 05:08.13 | p3nguin | The way it seems, I only need one that was used to activate an iPhone. |
| 05:08.15 | SeRi | You could buy a sim unlocker from china |
| 05:08.25 | p3nguin | It's a 3G, so it does not require the original SIM for this phone. |
| 05:08.28 | SeRi | its cheap like 5 dollars |
| 05:08.41 | SeRi | I see. |
| 05:08.42 | p3nguin | The 1 and 2 require the original SIM. |
| 05:08.58 | SeRi | Give me a few days. |
| 05:09.06 | SeRi | let me talk to some of my contacts in att |
| 05:09.08 | p3nguin | What would a SIM unlocker do for me? |
| 05:09.36 | SeRi | p3nguin, allow you to use the phone without a sim |
| 05:10.01 | p3nguin | dijib: I'm on the voipms echo test right now, and I'm not hearing any type of distortion or garbled sounds at all. It must have been something between me and zipdx. |
| 05:10.21 | p3nguin | Does it allow activation without a SIM? |
| 05:10.40 | p3nguin | The problem is that iTunes won't activate the phone without it. |
| 05:10.52 | SeRi | well never mind I keep thinking you have a 4 |
| 05:10.59 | SeRi | the sim unlocker only works with 4 |
| 05:11.23 | p3nguin | I'd gladly trade this one for a 4. |
| 05:11.42 | SeRi | give me a few dasy I can try and hunt an original sim card from an iphone 3GS from one of my contacts in att |
| 05:12.57 | p3nguin | I'd be surprised if there isn't an unused SIM from an iPhone floating around. |
| 05:13.05 | SeRi | I can make any promises but ill try |
| 05:13.13 | SeRi | cant* |
| 05:13.27 | SeRi | I am sure there is |
| 05:15.43 | SeRi | p3nguin, I bought one of this for 130.00 dollars http://www.portech.com.tw/p3-product1_1.asp?Pid=13 |
| 05:16.38 | SeRi | like new |
| 05:16.53 | SeRi | for my brother in PR. |
| 05:17.12 | SeRi | The idea is for him not to lose calls at his home/office |
| 05:17.23 | SeRi | PR is bad about cell signal |
| 05:18.13 | coppice | iphones with SIM lock? what a strange idea :-\ |
| 05:23.04 | p3nguin | seri: What do you think about something like this? http://www.ebay.com/itm/ws/eBayISAPI.dll?ViewItem&item=190577766131 |
| 05:25.09 | SeRi | p3nguin, They work very well. I use them in the v4. BUT the problem is that you have to make sure your iOS/Band version match |
| 05:25.17 | SeRi | do you know what version is load it? |
| 05:25.38 | p3nguin | I can't find out that information until I have a good SIM to allow iTunes to activate. |
| 05:26.56 | SeRi | and thats the problem. but hey its only two dollars |
| 05:27.05 | SeRi | actually 3 |
| 05:27.11 | SeRi | 1.99 shipping |
| 05:27.53 | SeRi | p3nguin, I trust this please more than ebay |
| 05:28.03 | SeRi | http://s.dealextreme.com/search/Unlock+Card+Chip+for+Apple+iPhone+3G |
| 05:28.09 | SeRi | I buy from them all the time |
| 05:28.16 | SeRi | check this one out |
| 05:28.33 | SeRi | http://www.dealextreme.com/p/universal-activation-sim-card-for-iphone-2g-3g-3gs-4-42595 |
| 05:29.42 | SeRi | lets hope you dont have iOS4 |
| 05:30.24 | SeRi | If you do I can tell you how to downgrade. |
| 05:31.02 | p3nguin | I'd like to have iOS 4. 3 doesn't have near the features and lots of apps I like don't work on 3. |
| 05:31.35 | SeRi | well get it unlocked first than you can move to iOS4 |
| 05:31.37 | SeRi | Is what I did |
| 05:31.42 | p3nguin | This universal activation card... you think that would do the trick for me? |
| 05:31.54 | p3nguin | Or should I just get an AT&T used card? |
| 05:31.58 | SeRi | If you are under 4 yes |
| 05:32.08 | SeRi | ether or. |
| 05:32.23 | p3nguin | Let's say I have an AT&T card that I can use to get iTunes to activate. |
| 05:32.27 | p3nguin | What's next? |
| 05:32.27 | SeRi | If you want to wait for me than wait before you purchase |
| 05:32.40 | SeRi | Next is the phun stuff |
| 05:32.42 | p3nguin | Do I need any other card to unlock carrier/network? |
| 05:32.54 | SeRi | no |
| 05:33.00 | SeRi | the rest is software base |
| 05:33.07 | SeRi | you need a mac or a windows machine |
| 05:33.10 | p3nguin | I don't need a turbo unlocker? |
| 05:33.14 | SeRi | a VM wont do. |
| 05:33.35 | p3nguin | I know how to jailbreak, it's the other stuff that I am not familiar with. |
| 05:34.12 | p3nguin | I don't know anything about unlocking carrier and whatnot. |
| 05:34.30 | p3nguin | I guess I just need to activate it and see what it will do. I'll go from there. |
| 05:34.45 | SeRi | You dont need no sim unlocker |
| 05:34.56 | SeRi | all you need is ultrasn0w for other sims to work |
| 05:35.09 | p3nguin | Should I prefer a used AT&T SIM or this universal activation SIM? |
| 05:35.14 | SeRi | you install ultrasn0w via cidia |
| 05:35.33 | p3nguin | How do you propose I'd get cydia on it? |
| 05:35.56 | SeRi | Let's say I have an AT&T card that I can use to get iTunes to activate |
| 05:35.56 | p3nguin | Typically you'd use something like ultrasn0w to put cydia on it. |
| 05:36.22 | SeRi | hu? mhhh nope |
| 05:36.31 | SeRi | ultrasn0w is an app IN cidia |
| 05:36.43 | p3nguin | So how do you propose I get cydia on it? |
| 05:36.54 | p3nguin | Some kind of Magic? |
| 05:37.03 | SeRi | d00d I am answering your questions |
| 05:37.05 | SeRi | Let's say I have an AT&T card that I can use to get iTunes to activate |
| 05:37.11 | SeRi | jailbrake it |
| 05:37.23 | SeRi | load ultrasn01 |
| 05:37.29 | SeRi | ultrasn0w |
| 05:37.31 | SeRi | done |
| 05:37.37 | SeRi | now you can use any carrier |
| 05:37.56 | p3nguin | This will be the third time I ask this: how do you propose I get cydia on it? |
| 05:39.16 | SeRi | fuck. LOL d00d I am answering the question you ask me "Let's say I have an AT&T card that I can use to get iTunes to activate" once you get your sim and have it activated it use the pawnage toold to jailbrake it and load cydia. |
| 05:39.44 | p3nguin | Maybe it's your lack of sleep or something, but cydia doesn't just magically appear. |
| 05:39.57 | SeRi | no is your lack of knowladge |
| 05:40.08 | SeRi | the pawnage tool loads cydia for you |
| 05:41.47 | SeRi | the pawnage tool has options to load in to the os. you can tell it what to load while is jailbraking your iphone... including cydia, sshd, etc... |
| 05:42.11 | SeRi | Unless something has change since my last unlock than pawnage does everything for you |
| 05:43.04 | p3nguin | pwnage tool should take care of much of it. |
| 05:43.30 | p3nguin | When I said you'd use something like ultrasn0w to put cydia on it, I was thinking redsn0w. |
| 05:43.51 | p3nguin | redsn0w would provide the jailbreak and cydia. |
| 05:43.53 | SeRi | ah! |
| 05:44.00 | p3nguin | Then you could install the ultrasn0w app. |
| 05:44.02 | SeRi | yes yes yes :) |
| 05:44.34 | SeRi | many tools out there that can do the job. |
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| 05:44.59 | p3nguin | But the main thing is knowing if it makes a difference if I get the universal activation sim or a used at&t sim. |
| 05:45.30 | p3nguin | If they will both be equally effective and useful, I'll probably go with the at&t used sim. |
| 05:46.11 | SeRi | well with the ATT you have a better chance that it will work even with iOS4.... where the universal only works with 3.x and under.... |
| 05:50.01 | SeRi | p3nguin, Ill call my contact tomorrow and ill let you know probably by noon or so |
| 05:56.35 | p3nguin | I'm curious if it has to be a SIM from an iPhone, or could it simply by any AT&T phone. |
| 05:56.46 | p3nguin | s/by/be/ |
| 05:58.34 | SeRi | Not sure. |
| 05:58.47 | SeRi | I dont think is iphone specific |
| 05:58.56 | SeRi | well sims at least |
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| 06:26.21 | sbk[1] | any body there? |
| 06:26.30 | sbk[1] | i'm new to irc |
| 06:26.40 | sbk[1] | and i wanna join asterisk irc |
| 06:26.41 | sbk[1] | :P |
| 06:30.58 | ChannelZ | too bad, you did. |
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| 07:17.23 | LiENUS | i was curious, has anyone used something like this http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-analog-gateways/gxw410x as an analog interface for a fxo? |
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| 07:21.25 | k3asd` | hi |
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| 07:21.35 | sbk[1] | hi |
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| 07:45.56 | Nunners | Morning all.... voicemail question. I've setup a php mail script as the mailcmd (mailcmd=php /etc/asterisk/voicemail.php) restarted, however when I verbose 5 the console, it's now calling the file - should it be this simple?! |
| 07:50.14 | dym | Nunners: well if the file contains everything - then yeah |
| 07:50.35 | dym | Nunners: Also see - http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf |
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| 08:59.57 | AlecTaylor | hi |
| 09:00.31 | EmleyMoor | When using ChanSpy's "barge" options, what is the difference between that and (a) spy (b) whisper) |
| 09:00.34 | EmleyMoor | ? |
| 09:00.38 | AlecTaylor | If I were to teach asterisk or freeswitch in a classroom environment, is there someway I can emulate and script emulation of users calling in/receiving calls, transfering calls &etc? |
| 09:04.44 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 09:06.14 | ChannelZ | EmleyMoor: Wisper lets you talk to only one side |
| 09:09.06 | EmleyMoor | ChannelZ: So spy = listen only, whisper = talk to my side, barge = effectively become part of a 3-way? |
| 09:09.34 | ChannelZ | Pretty much yes |
| 09:10.29 | EmleyMoor | Thought so - will try it out next time the kind of call I listen to happens - have to be close to the originating phone to be heard other-side at present but they are actually meant to be for both of us |
| 09:10.49 | EmleyMoor | (or to the phone at my end, for a received call) |
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| 09:12.49 | AlecTaylor | pfft, this channel is dead. Contacted mailing-list... |
| 09:13.07 | ChannelZ | AlecTaylor: Use softphones |
| 09:15.17 | ChannelZ | And people do sleep. |
| 09:15.49 | AlecTaylor | People don't sleep! |
| 09:15.52 | AlecTaylor | Don't be ridiculous :P |
| 09:16.08 | AlecTaylor | Are softphones scriptable? |
| 09:16.24 | AlecTaylor | Anyways, wrote a more detailed explanation on what I'm trying to accomplish on the mailing-list |
| 09:16.35 | ChannelZ | I'm sure there are ones that are |
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| 09:18.33 | AlecTaylor | link? |
| 09:18.58 | ChannelZ | google.com |
| 09:19.48 | ChannelZ | ~softphones |
| 09:19.58 | ChannelZ | hmmm |
| 09:20.27 | ChannelZ | ~softphone |
| 09:20.28 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
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| 12:02.23 | devil_evoxxx | hi all guys |
| 12:18.14 | devil_evoxxx | someone can explain me how cdr "rating" could work? |
| 12:19.04 | devil_evoxxx | i have an account associated with two, or three number |
| 12:20.13 | devil_evoxxx | and i need to calculate the effective cost, specially in case of two / three concurrent calls |
| 12:20.17 | devil_evoxxx | some idea? |
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| 12:24.06 | jsjc | I wonder if anybody could show me an example of an if condition on dialplan such us if this dahdi DIAL is not succesful for anyreason then do this other DIAL. |
| 12:28.21 | EmleyMoor | jsjc: The only reason I can think of that a DAHDI dial might actually fail is that there isn't a channel available... that doesn't need a condition and just presses on. |
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| 13:23.57 | wonderworld | hey, in conf bridge 10, is the config file reread before creating a new conf bridge instance, or is it read only at asterisk start? |
| 13:30.11 | kaldemar | wonderworld: at start/reload |
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| 13:40.56 | wonderworld | conf bridge 10 really looks nice. have there been reports already if it's stable with many users? |
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| 13:49.33 | devil_evoxxx | hi, someone use cdr over radius here? |
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| 14:04.26 | cusco | hi |
| 14:05.27 | devil_evoxxx | hi |
| 14:06.24 | devil_evoxxx | irroot: hi |
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| 14:08.15 | cusco | I'm still having some nat issues |
| 14:08.21 | cusco | can't find out why |
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| 14:13.55 | devil_evoxxx | ast version? |
| 14:14.05 | devil_evoxxx | someone say where is the log of radiusclient cdr using asterisk? |
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| 14:30.26 | nedos | hey guys, I'm trying to get my head around the TLS encryption in asterisk? Does it only support the TLS style auth that you get with OpenVPN where you have client and server certs? I wanted to encrypt my sip traffic originating from my iphone etc... |
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| 14:38.51 | cusco | [TK]D-Fender: seems that I still hae nat issues and I can't figure out why |
| 14:38.58 | cusco | I see: Sending to 88.157.128.26:5060 (no NAT) |
| 14:40.33 | cusco | http://paste.debian.net/136735/ here is the full sip debug |
| 14:40.53 | cusco | I can't figure what is wrong.. |
| 14:41.09 | cusco | in there I have 2 softphones at work, registering in asterisk at home |
| 14:41.20 | cusco | if I'm at home, it works, other people can hear me |
| 14:42.31 | devil_evoxxx | connection tracking |
| 14:42.33 | devil_evoxxx | table? |
| 14:43.33 | cusco | I looked at wireshark |
| 14:45.00 | cusco | rtp seems to gom from my local IP to asterisk external ip |
| 15:08.56 | [TK]D-Fender | cusco: Check your firewalls. So far it looks OK |
| 15:16.18 | cusco | that 192.168.1.3 in there.. is it ok? |
| 15:16.29 | cusco | at home I have DMZ set to asterisk box |
| 15:16.50 | [TK]D-Fender | cusco: Set your peers as nat=yes |
| 15:16.55 | cusco | they are |
| 15:17.05 | [TK]D-Fender | cusco: even though they do seem to be posting the right Contact: header info. |
| 15:17.22 | [TK]D-Fender | <--- Transmitting (no NAT) to 88.157.128.26:5060 ---> |
| 15:17.25 | [TK]D-Fender | Apparently not |
| 15:17.47 | [TK]D-Fender | eliably Transmitting (no NAT) to 88.157.128.26:5060: |
| 15:18.13 | cusco | yes that is what I find weird |
| 15:18.28 | cusco | does it need some order on the peer flags? |
| 15:18.33 | [TK]D-Fender | nat=yes" <- |
| 15:18.54 | [TK]D-Fender | go fix this. From what * is reporting your assertion seem incorrect |
| 15:19.07 | cusco | [TK]D-Fender: http://paste.debian.net/136747/ |
| 15:19.29 | cusco | ah |
| 15:19.30 | [TK]D-Fender | nat=route <- doesn't say "no" |
| 15:19.31 | cusco | dooh, |
| 15:19.41 | cusco | yes I changed it just to try |
| 15:19.58 | cusco | but still.. allow me to set it to yes, and reproduce |
| 15:21.22 | cusco | ok I set it to nat=yes on both peers, and sip reload on cli |
| 15:21.27 | cusco | http://paste.debian.net/136749/ |
| 15:21.47 | [TK]D-Fender | cusco: Srtill no audio? |
| 15:21.57 | cusco | no |
| 15:22.13 | cusco | but if doing it at home (home network) it works |
| 15:23.21 | cusco | sip.conf: http://paste.debian.net/136750/ |
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| 16:51.52 | budman_mtl | hello all |
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| 16:53.23 | budman_mtl | i'm new here, and i wonder i asterix is the tool i need to accomplish a job... i want to make a personal anac for my technician, they would have to dial in, enter a passcode, and then get all the info for the line they calling from, i'm i looking at the right place ?? |
| 17:05.30 | [TK]D-Fender | budman_mtl: "all info" is the grey area. There is caller ID. What more are you expecting? |
| 17:05.42 | [TK]D-Fender | budman_mtl: because that's generally all a phone system knows |
| 17:06.07 | [TK]D-Fender | budman_mtl: Everything else that could possibly be looked up is an external lookup into some other database. This is up to you |
| 17:10.57 | budman_mtl | i got database allready, thxs for the answer so going with asterix i will be able to do my job |
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| 17:11.27 | budman_mtl | i guess i have long hours of learning the ael |
| 17:11.30 | dynamite | reeee |
| 17:12.49 | WIMPy | budman_mtl: If what you described is all you need, I think a single line will be enough. But you should look at AGI. |
| 17:13.21 | budman_mtl | will do thxs, i,m totl neebee w asterix |
| 17:14.20 | WIMPy | More detailed questins lead to better answers. |
| 17:14.37 | irroot | yo there WIMPy |
| 17:15.10 | irroot | found the problem with the timeouts some sort of race condition on starting lcr |
| 17:15.10 | WIMPy | Hi irroot. Did you get any feedback on the version number issue, yet? |
| 17:15.30 | budman_mtl | i will start by doing my homework, and trust me i will come back with tons of Q lol;) |
| 17:15.40 | p3nguin | budman_mtl: It's asteriSK. |
| 17:15.56 | WIMPy | ~book |
| 17:15.56 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 17:16.00 | irroot | WIMPy the version number is a problem only on the 1.8 branch and i did raise a discusion ill put forward a patch |
| 17:16.03 | WIMPy | budman_mtl: Start there ^^ |
| 17:16.18 | irroot | WIMPy if you have .version set then there is no problem |
| 17:16.23 | budman_mtl | BIG THANX ! |
| 17:17.20 | WIMPy | irroot: Well, it more of a general issue. With SVN versions you dan't have any relation to the release versions. That's suboptimal. |
| 17:17.39 | WIMPy | What kind of race condition did you find? I don't think I cam across that. |
| 17:18.08 | irroot | i run my own distro to start with so that 1/2 problem |
| 17:18.10 | WIMPy | But I think I saw an issue with multiple terminals on one port. Need to investigate. |
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| 17:18.33 | [TK]D-Fender | bugI recommend against AEL |
| 17:18.35 | irroot | on boot up the USB/PCI is probed and drivers loaded |
| 17:18.37 | WIMPy | I Don't use any distro-specific stuff. |
| 17:18.47 | irroot | real early on |
| 17:18.50 | [TK]D-Fender | budman_mtl: * |
| 17:19.06 | irroot | then DSP is loaded if it sees /dev/mISDNtimer |
| 17:19.59 | budman_mtl | ok TK |
| 17:20.09 | irroot | starting lcr early or even at the end of the rc.xxxx files after a call to the line ie out / in it would time out and loop forever |
| 17:20.24 | irroot | unloading / loading port would resolve all ills |
| 17:21.11 | WIMPy | irroot: Strange I have used my netbook for testing and I heve even started LCR before plugging in the adaptors without issues. |
| 17:21.41 | WIMPy | Just plug it is and type interface in the admin console. |
| 17:21.42 | irroot | but starting it after init completes all is well have a few sites running it now live |
| 17:22.30 | irroot | it had me stumped |
| 17:23.02 | WIMPy | Maybe it's because you have to reload interfaces that it works. |
| 17:23.33 | irroot | i have a cron job that runs periodically to check services its starts from there no issues no reload and rock solid |
| 17:23.49 | WIMPy | However I've been told by Andreas that you don't need to do that when you name the interfaces. |
| 17:24.15 | irroot | so you think puting a name on iterface could help in the config ? |
| 17:24.57 | irroot | ill test it sometime |
| 17:25.03 | WIMPy | I haven't tried that. |
| 17:25.13 | WIMPy | But it might be related. |
| 17:25.27 | irroot | thx really appreciate your input |
| 17:26.22 | WIMPy | And I heven't trieD to configure Asterisk as interface, either. |
| 17:27.09 | irroot | i need NT and TE mode and the digium drivers dont seem to support NT mode for ptmp i want to be "inline" passing calls via asterisk to existing PBX for recording / queue / ivr / .... featutres |
| 17:28.00 | WIMPy | dahdi does support NT-ptmp |
| 17:28.56 | WIMPy | And dahdi has one big advantage: It supports ECT. But otherwise I prefer LCR, especially because of the screening options. |
| 17:29.32 | WIMPy | With dahdi you have no way to check CallerID. It always uses whatever the user supplied. |
| 17:29.50 | irroot | # NOTE: When using BRI channels in asterisk, use the bri_cpe, bri_net, or |
| 17:29.52 | irroot | # bri_cpe_ptmp (for point to multipoint mode). libpri does not currently |
| 17:29.55 | irroot | # support point to multipoint when in NT mode. Otherwise, the bearer channel |
| 17:29.56 | irroot | # are configured identically to other DAHDI channels. |
| 17:30.15 | WIMPy | That's outdated. |
| 17:30.21 | irroot | ah ok thx |
| 17:30.22 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN |
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| 17:31.04 | irroot | i prefer LCR myself as it allows me to mix USB and PCI have a few customers with USB+HFCMulti |
| 17:31.29 | WIMPy | Yes, no USB support in dahdi. |
| 17:32.59 | WIMPy | But ECT really is a huge point. |
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| 17:35.43 | WIMPy | Ouch. My X crashed :-( |
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| 17:37.25 | irroot | that is a bummer |
| 17:38.48 | WIMPy | Yes, especially when you are about to place a bid on ebay. |
| 18:03.11 | irroot | WIMPy you could add Sangoma Wanpipe SMG and native DAHDI ISDN drivers |
| 18:07.05 | WIMPy | Doesn't the Sangoma stuff interface to dahdi? |
| 18:07.12 | WIMPy | And what are you missing about dahdi? |
| 18:08.22 | WIMPy | I should probably add more about capi. But the trouble there is that it doesn't only depend on chan_capi but the specific capi implementation as well. |
| 18:11.48 | irroot | WIMPy the sangoma stuff only in latest release interfaces BRI there was there own stack previously |
| 18:12.39 | WIMPy | I was under the impression that the wanpipe stuff acted like a hadrware driver to dahdi. |
| 18:13.12 | WIMPy | But I didn't come across any Sangoma hardware, yet. |
| 18:13.47 | WIMPy | And there isn't much hope that any will appear at a "nice for testing" price, I guess. |
| 18:13.48 | irroot | they had a HFC card A500 that needed a seperate BRI stack and chan_woomera |
| 18:14.18 | irroot | its very expensive but did have hands on one for a while |
| 18:14.43 | irroot | got a good rep with the importers they let me play with it for feedback and training them |
| 18:15.51 | WIMPy | Oh, that's a VOIP thing? Interesting. |
| 18:17.45 | WIMPy | Do I get that right? The Card comes with a H.323 gateway that then converts to woomera and goes in to Asterisk via chan_woomera??? |
| 18:19.43 | WIMPy | Or maybe better s/converts/tunnels/ |
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| 18:22.41 | irroot | WIMPy womera is like chan_lcr in many ways its a socket system that uses rtp |
| 18:23.20 | irroot | so it sends "messages" to the "media" gateway to establish the call |
| 18:24.22 | WIMPy | Yes, but I read UDP there. And it seems to tunnel existing VOIP protocols like H.323 or SIP. |
| 18:25.46 | irroot | WIMPy yeah in a real watered down way |
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| 18:29.42 | WIMPy | There doesn't seem to be much information about woomera available. |
| 18:30.19 | irroot | no there is not but with the latest bits the BRI is native |
| 18:30.42 | WIMPy | native = dahdi? |
| 18:30.42 | irroot | there SMG [sangoma media gateway] is still available |
| 18:30.57 | irroot | yip Sangoma+DAHDI+libpri |
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| 18:31.35 | WIMPy | Ok, so it kust falls under dahdi now |
| 18:32.13 | irroot | but requires the wanrouter configured first ... |
| 18:32.42 | WIMPy | yes |
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| 20:25.03 | Greenlight | Hiya, I'm having some issues with jitterbuffer over an IAX2 trunk. Whenever I enable the jb I get garbled and choppy audio, without the jb it's much better but there are still times when it's choppy. The is plenty bandwidth and I've tried using gsm and g729. Could it be that the remote end is transcoding to alaw and this causes issues with the jb? |
| 20:26.02 | WIMPy | What tech are the endpoints using? |
| 20:26.29 | WIMPy | You should only enable jb for the endpints, not for trunks, usually. |
| 20:27.12 | Greenlight | Well the calls will be going into a confbridge on the remote asterisk box, which is why i've been trying to enable jb as we were getting some choppy audio |
| 20:27.17 | Greenlight | The setup is: |
| 20:28.00 | Greenlight | SIP softphone <--LAN--> Asterisk 1 <--IAX Trunk WAN--> Asterisk 2 <--IAX Trunk WAN --> ITSP |
| 20:29.19 | Greenlight | The calls where we were having issues were going into a ConfBridge on Asterisk 2 |
| 20:29.21 | WIMPy | In that setup I'd recommend to only enable JB on the SIP part and not for any of the IAX connections. |
| 20:29.56 | Greenlight | So for the SIP part, you mean just the jitterbuffer built into the softphone? |
| 20:29.57 | WIMPy | What timer do you use? 'timer test' |
| 20:30.24 | Greenlight | Asterisk 2: |
| 20:30.25 | Greenlight | Attempting to test a timer with 50 ticks per second. |
| 20:30.26 | Greenlight | Using the 'timerfd' timing module for this test. |
| 20:30.26 | Greenlight | It has been 1000 milliseconds, and we got 50 timer ticks |
| 20:30.34 | WIMPy | No, you can enable it on Asterisk as well. |
| 20:30.40 | Greenlight | Attempting to test a timer with 50 ticks per second. |
| 20:30.40 | Greenlight | Using the 'DAHDI' timing module for this test. |
| 20:30.40 | Greenlight | It has been 1019 milliseconds, and we got 51 timer ticks |
| 20:30.57 | WIMPy | You might want to try to install dahdi and use that as timing source. |
| 20:31.19 | WIMPy | And I've never seen it overshoot before. |
| 20:31.30 | WIMPy | Are you using virtualisation? |
| 20:31.32 | Greenlight | All my DAHDI installs do that |
| 20:31.42 | Greenlight | And mosts pastes i'v seen on forums |
| 20:31.48 | Greenlight | 1019ms - though it was just standard |
| 20:31.54 | Greenlight | It's using a sagmoa usb timer |
| 20:32.37 | WIMPy | You just pased 'timerfd'. That has caused issues for some. |
| 20:32.54 | WIMPy | s/passed/pasted/ |
| 20:33.14 | Greenlight | Think pthreads may be safer? |
| 20:33.54 | WIMPy | The best option is dahdi. pthreads should be last choice. |
| 20:35.21 | Greenlight | Have got a 1000hz kernal, HPET clock source - and other things all seem okay on that server, its just when I enable jb that I have issues |
| 20:36.05 | Greenlight | I used to use DADHI timer and MeetMe conf, but am running a custom kernel now and I've not got the sources for it so can't compile dahdi |
| 20:36.22 | WIMPy | I once forcedenable JB for IAX to get statistics, but that definitely made things worse. |
| 20:36.47 | Greenlight | Yea - thats what I mean it just seems to increase the jitter if anything |
| 20:37.17 | WIMPy | I don't think it can be a good thing to have multiple jitterbuffers anyway. |
| 20:37.38 | WIMPy | It can only add delay. |
| 20:37.44 | WIMPy | And that's not a good thing at all. |
| 20:37.50 | Greenlight | True - but it shouldn't add jitter |
| 20:38.06 | WIMPy | No, it shouldn't. |
| 20:38.10 | Greenlight | If i have jb disabled on all iax trunks can i still use the JITTERBUFFER dialplan appliaction ? |
| 20:38.38 | Greenlight | My thinking is that I can't, and thats why im no able to dejitter audio into ConfBridge |
| 20:39.33 | WIMPy | I've never tried that one, but I don;t see what use it would be if it wasn't independant of the channel it uses. |
| 20:39.42 | Greenlight | audio from the istp is perfect when into the confbridge (4ms ping, no jitter), but audio from the other asterisk box is quite choppy. |
| 20:39.51 | Greenlight | Yea, good point |
| 20:40.21 | Greenlight | I read somewhere about transcoding and jitterbuffering having some issues but couldn't see an absolute answer to it |
| 20:41.16 | WIMPy | Transcoding can also involve different paketisation times. |
| 20:41.35 | WIMPy | If the timing is the same, there shouldn't be any effect. |
| 20:41.54 | Greenlight | iirc alaw and g729 are both 20ms frames? |
| 20:41.55 | WIMPy | But having different timing might be worth avoiding. |
| 20:42.16 | WIMPy | G.729 defaults to 30ms, IIRC. |
| 20:42.32 | Greenlight | AH - what about GSM? |
| 20:43.36 | WIMPy | The low bandwidth codecs tend to use larger frames as the RTP overhead would make the lower codec bitrate pretty pointless otherwise. |
| 20:44.04 | WIMPy | No, gsm defaults to 20. |
| 20:44.24 | WIMPy | And so does G.729. |
| 20:44.42 | WIMPy | But Phones might have a different default. |
| 20:44.42 | Greenlight | Yea - thats what makes IAX trunks so attractive an option in low bandwidth enviroments |
| 20:44.53 | WIMPy | Indeed |
| 20:45.32 | Greenlight | Just can't for the life of me work out what's causing this choppy audio |
| 20:46.23 | WIMPy | It might even be related to the version. Are you using the old or the new jitterbuffer? |
| 20:47.02 | WIMPy | I read the adaptive one took some time to become really stable. |
| 20:47.12 | Greenlight | Whatever the default is on iax trunks in 1.8.7 and SVN-trunk-r337855M |
| 20:48.17 | Greenlight | Yea - I did read some info but most of it was years ago, and newer info suggested that it was all nice and working now |
| 20:48.46 | WIMPy | No idea what the default is, but chances are the adaptive version is working correctly on 1.8.7. |
| 20:48.55 | Greenlight | Yea that's what I presumed |
| 20:49.57 | Greenlight | Anyone else had any issues (good or bad) using IAX jitterbuffer? |
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| 21:04.04 | Greenlight | I think I've found a solution which seems to work, or at least vastly improve, the issues I was having with jitter |
| 21:04.50 | Greenlight | I've left jb disabled in all the conf files, but when I connect the user into the conference if I use local/XXXX@YYY/nj it seems to clear things up |
| 21:05.18 | Greenlight | My only guess is that it's dejittering prior to transcoding and that must help in some way |
| 21:06.05 | WIMPy | I was about to sugest a local channel, but I thought that the JITTERBUFFER function would do the same, even if only in one direction. |
| 21:06.45 | WIMPy | Which would be exactely right, but maybe someone will enlighten us some time, what it really does. |
| 21:08.23 | Greenlight | Indeed - thats why my thinking is that it's to do with when the dejitter happens - before or after the transcode, what must be effecting it |
| 21:09.56 | Greenlight | Well time for some more testing then got to recode the local channel stuff into my dialler application when it originates calls out of the confbridge |
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| 21:37.12 | LiENUS | anyone know how hp multi function print scan fax machines handle networked faxing? |
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| 23:17.34 | SeRi|afk | guys whats the equivalent for insecure=very in 1.8? |
| 23:24.03 | [TK]D-Fender | insecure=port,invite |
| 23:28.21 | dym | BTW: Fixed the permanent conference bridge: exten => 334,1,Dial(SIP/astconf@sip.openroot.de) |
| 23:28.25 | dym | for those interested |
| 23:28.52 | LiENUS | is it possible to integrate asterisk with a cups based fax machine? |
| 23:29.10 | LiENUS | the faxes are sent by printing to cups |
| 23:32.02 | LiENUS | was wondering if theres some way to have asterisk accept t.38 faxes and print them to the fax machine so they fax out |
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| 23:38.49 | SeRi|afk | Thanks [TK]D-Fender |
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| 23:48.03 | ldiamond | Anyone know a good sip trunk provider that provide DID with incoming and outgoing SMS? |
| 23:48.17 | ldiamond | Anveo.com does it but it's complete junk. |
| 23:50.09 | ChannelZ | I think Vitelity does |
| 23:51.38 | Korolev | and you really need incoming sms |
| 23:51.47 | Korolev | outgoing sms is pointless in the US and Canada |
| 23:51.51 | ldiamond | yes |
| 23:51.57 | ldiamond | sadly everyone relies on it |
| 23:51.59 | Korolev | every mobile carrier has email gateways |
| 23:52.12 | ldiamond | but I have to receive. |
| 23:52.16 | Korolev | i meant you really only need* |
| 23:52.30 | ldiamond | well, outgoing is a given if a voip provider has incoming |
| 23:52.35 | ldiamond | usually they have an sms gateway |
| 23:52.42 | Korolev | sure, but they will charge your for them |
| 23:52.55 | ldiamond | but it will show up as my number on the target |
| 23:53.27 | ChannelZ | http://www.vitelity.com/services/sms |
| 23:54.29 | ldiamond | I'm in Canada though, they say "US" not "Canada". I'll give it a shot though |
| 23:54.43 | ldiamond | err 35$ minimum to try... |
| 23:55.44 | ldiamond | voip.ms will be adding SMS support... they're taking forever! |
| 23:57.57 | ldiamond | or Google voice should be available in canada |
| 23:58.57 | ldiamond | any other alternatives? |
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